D4 Dialer, Predictive Dialer, VICIDIAL, Asterisk based dialer, Clear voice D4Codec, Ahmedabad, Gujarat, India, Dialer with all features

D4Dialer - Company Profile (A user friendly predictive dialer with unbeatable rates and uncompromized service)

2010-03-03T18:57:00.002+05:30

Respected Sir / Madam

We, D4Dialer – http://www.d4dialer.com (A unit of Solution Empire since 1997), provide international and domestic call center setup on predictive dialer with world wide asterisk technology as backend and using most user friendly CRM Vicidial. We customized inbound/outbound dialer as per client requirement with their branding also. We customize vicidial, asterisk, trixbox, FreePBX etc. large scale voip based solution.

We provide premises based and web based dialer server on linux platform with asterisk technology. The Major features of Dialer provided by us as below.

Major Features Of Dialer

ü Inbound, Outbound and Blended call handling

ü Outbound agent-controlled, broadcast and predictive dialing

ü Web-based agent and administrative interfaces

ü Integrated call recording

ü Three-Way calling within the agent application

ü Ability to use standard Telco lines and VOIP trunks

ü Multiple campaigns and lead-lists are possible

ü Real-time campaign display screens

ü Ability to automatically record all calls

ü Faster hangup and dispositioning of calls with one key press (HotKeys)

ü Lowest Drop Ratio

ü And much more than your desire……………..

Like other suppliers, we talk about quality & service, but-more important-we take steps to provide quality & service that is measurably higher, a cut above the competition.

Why Choose D4Dialer :

Fully Technical Support
Immediate Solution
Qualified Technical Team
User Friendly Technology

Since 1997 in Technology
100% Reliable Work
Complete Technology Solution under ONE Roof
Affordable Rates

Other Services Provided By our Parent Company Solution Empire (http://www.solutionempire.com)

Website Designing

ü Static Website Designing

ü Dynamic Website Designing

ü Php Website Designing

ü Flash Website Designing

ü HTML Business Catalogue

E Commerce

ü Shopping Cart

ü Jewelry Website

ü Hotel Directory with Reservation System

ü Travel Portal with Reservation System

Website Maintenance
Newspaper Website
Content Management System (CMS)
Customer Relationship Management (CRM)
Search Engine Optimization
Payment gateway Integration
Paypal Integration
Open source Customization
Domain Name Registration
Web Hosting – Linux
Total Web Solution

At Solution Empire (D4Dialer), we understand that Business Operations today demand extremely high operational productivity and administrative efficiency with quick turnaround times. The key to success is in transforming and optimizing processes to achieve these results and at the same time, allow businesses to focus on their core activities.

We look forward to positive business response with you.

Thanks & Regards.

Deepak Jain

(+91-95588 16181)

D4DIALER

Mobile : +91 - 95588 16181 , Email : dialer@d4dialer.com , Website : www.d4dialer.com

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2009-12-28T21:43:00.003+05:30

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To develop your website please visit
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Call on +91-9558816181

Email : ceo.solutionempire@gmail.com

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2009-12-13T20:12:00.014+05:30

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FEATURE LIST :

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 AGPLv2 licensed, with no software licensing cost

Full Feature 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/verifier 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
Automtically dial unlimited numbers per customer until you get an answer
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 specified status 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 of call across multiple inbound or outbound Asterisk servers is possible
Agent phone login balancing and failover across multiple DIALER servers
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
Custom Music-On-Hold and agent alert sound for inbound calls
Estimated hold time, place in line, overflow queues and several other inbound-only features
Skills-based ranking and call routing per inbound group(queues) and campaign
Queue Prioritization per campaign and inbound group
Single agent call queueing
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
Agent shift enforcemant by day and time, defined per user group
Full QueueMetrics-compatible call logging, inbound and outbound
Several Vtier integration features: user-sync, account-sync, data interconnection
Full integration with Sangoma Call Progress Detection(CDP) for better Answering Machine Detection(AMD)
Multi-function web-based agent API allowing for control of agent sessions including click-to-dial outside of the agent screen
Lead import web-based API
Web-based data export utilities
Separate Time-clock application to track user work time
Web-based administration

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If you face any problem or any query please feel free to contact us on d4dialer@gmail.com . We charge consultancy to solve the problem.

For clear voice clarity please buy D4CODEC from us.

Thanks & Regards.

D4 Dialer.
110B Nirman house, b/h Times of india,
Ashram road, Ahemdabad : 380009
Mob : 0-9558816181
email : d4dialer@gmail.com

solutionbposervices@gmail.com

www.d4dialer.com

If you find any home based work for individuals please visit http://www.netvyapar.com/mailer

Note: Unlimited VOIP minutes Terms as mentioned on our online VOIP partner.

Total Feature List of Predictive Dialer (VICIDIAL) Customzied by D4DIALER at FREE of COST

2009-12-08T13:11:00.010+05:30

MAJOR VICIDIAL FEATURES Customized by D4DIALER:

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 AGPLv2 licensed, with no software licensing cost

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VOICE clarity from US

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Full Features List customized by D4DIALER:

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/verifier 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
Automtically dial unlimited numbers per customer until you get an answer
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 specified status 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 of call across multiple inbound or outbound Asterisk servers is possible
Agent phone login balancing and failover across multiple ViciDial servers
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
Custom Music-On-Hold and agent alert sound for inbound calls
Estimated hold time, place in line, overflow queues and several other inbound-only features
Skills-based ranking and call routing per inbound group(queues) and campaign
Queue Prioritization per campaign and inbound group
Single agent call queueing
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
Agent shift enforcemant by day and time, defined per user group
Full QueueMetrics-compatible call logging, inbound and outbound
Several Vtier integration features: user-sync, account-sync, data interconnection
Full integration with Sangoma Call Progress Detection(CDP) for better Answering Machine Detection(AMD)
Multi-function web-based agent API allowing for control of agent sessions including click-to-dial outside of the agent screen
Lead import web-based API
Web-based data export utilities
Separate Time-clock application to track user work time
Web-based administration
DID, phone and carrier trunk provisioning through the web interface
Client web-app web pages available in English, Spanish, Greek, German, French, Italian, Polish, Portuguese, Brazillian Portuguese, Slovak, Russian and Dutch.
Admin web pages available in English, Spanish, Greek, German, Italian and French.

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BlogUniverse

Screenshot of Dialer - Great Look for agents and admin by D4DIALER

2009-12-05T13:48:00.004+05:30

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CODEC For CLEAR VOICE clarity from US

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CODEC

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We Provide Consultancy for VICIDIAL to improve your voice quality, clarity and stable work.

D4 Dialer

(Unit of SOLUTION EMPIRE)
301, Sarthak – II, Nr. Liberty Showroom,
B/h Swastik Bus Stand, Swastik Char Rasta,
C G Road, Ahmedabad : 380009 (IN)
Mobile : 0 - 95588 16181
Email : d4dialer@gmail.com

www.d4dialer.com

Download Link for VICIDIAL and ASTERISK

2009-12-05T13:44:00.007+05:30

download link location

http://sourceforge.net/projects/astguiclient/files/astguiclient_snapshot/astguiclient_2.0.5.1rc1/astguiclient_2.0.5.1rc1.zip/download

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VICIDIAL Scratch Installation Information available on eflo.net as opensource dialer

2009-12-05T13:43:00.005+05:30

Asterisk/astguiclient install from scratch. v.2.0.5 2009-04-03

By the VICIDIAL group, You can consult for install on d4dialer@gmail.com

**** IMPORTANT - In order for vicidial/astguiclient to function correctly please

read the REQUIREMENTS.txt for a minimum requirements list. ***

End-user Manuals for Agents and Managers are available from http://www.eflo.net

This document is meant to be a very in-depth step-by-step explanation of

installing the Asterisk open-source PBX on a Linux system and also installing

the astGUIclient suite. The instructions will assume starting from nothing and

will try to give several side step instructions to account for some differences

in choices of hardware and software.

The actual installation that I am doing as I write these instructions will be on

the following hardware:

- Pentium 3 500MHz

- Intel motherboard D815BN

- 256MB PC133 RAM

- 80GB IBM deskstar 7200RPM Hard Drive

- Digium Wildcard Single Span T1 Card T100P

- 2U rackmount case with 250W power supply

- Phone hardware will be a Grandstream BT102 and a Sipura SPA-2000 because they

are so cheap and readily available

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All of these parts, aside from the Digium card and the two SIP VOIP devices,

were purchased from ebay and the entire package(with the two VOIP devices and

all server hardware included) cost me about $1100 to put together including the

phones and Digium adapter.

We have many other Asterisk servers at our main office, but this one can be

experimented with easily because it was so cheap to make and has a relatively

small capacity when compared with a multi-processor server with a quad span T1

card.

This is our test Asterisk server and functions well for a dozen or so extensions

in use if it were to be used in production. A size that is optimal for many

small offices operating with a fractional data/voice T1 for instance.

For hardware you can use almost any Pentium-class processor(PII, PIII, Athlon,

Xeon, etc), and you can use any digium telco interface card. Both of these

choices will determine what the capacity of your Asterisk server will be. If you

want to do simple IVR or conference calling and a few extensions, then a PIII

with a single Digium T1 card will work just fine for you. If you want to use the

VICIDIAL application, you will want to get as high-powered of a machine as you

can afford and get a digium quad-span T1 card.

The following is assumed for these installation procedures:

- You have access to a CD burner and 3 blank CDs

- You have some sort of broadband internet connection

- You understand basic linux commands and can use a file editor like vi

- You have all of the necessary hardware:

- a pentium-class computer

- a digium telco interface card with appropriate telco lines

- at least 1 SIP VOIP device

- a Local Area Network(LAN) with extra ports enough for the new server

and the number of phones you want

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PHASE 1: INSTALLING AN OPERATING SYSTEM

This installation will be using Slackware 11 for the linux distribution,

Slackware 12.X will also work with these instructions. There are several easier

linux distributions and there are others that are more popular, but Slackware is

a nice non-commercial distro that has been around for a long time and proven

itself to be a very uncluttered and stable platform for development.

1. Go to http://www.ultimatebootcd.com/ , download the latest bootcd and

burn it to a CD. This will be needed to partition the hard drive prior to

installation of Slackware linux. The latest version as of this writing is 4.1

(If you have problems with your hardware booting some of the utilities with 4.1

I suggest trying 1.7, that version has older utilities, but still gets the job

done and works on every machine I've tried it on).

2. Insert the ultimatebootcd you just burned into your CDROM drive and boot

to it. You will select "filesystem utilities" and then "XFDISK"

3. Select any old partitions and delete them and then create 2 new

partitions:

- 70000 MB, select yes to validate, change partition type to "Linux Native"

- 3332 MB, select yes to validate, change partition type to "Linux Swap"

- press F3 to exit and let it do it's thing, this will take an hour or so.

4. Go to http://www.slackware.com/getslack/ to download Slackware linux.

The most recent release we recommend is 11.0. This release fits on 3 CDs

or 1 DVD. Download both installation disks from any close server listed on the

download page and burn them both to CDs.

5. Insert Disk 1 of the Slackware installation CD and boot your computer.

If you have a simple computer with just an IDE drive just hit enter at the boot:

prompt. If you have other hard drive adapters(SCSI/RAID/SATA/etc..) you will

need to look at the Slackware installation help page to determine what boot

image you will need to use to install Slackware correctly.

6. Login as root and type "setup" at the prompt to go to the setup menu.

7. Go to ADDSWAP and hit Enter

8. Select the swap partition you just created and hit Yes, The swap

partition will then be formatted

9. Select the root partiton you just created as Linux Native and hit

Select, then select "ext3" for the file system, then select 4096 for the inodes

and the root partition will then be formatted

10. Select "Install Slackware from a CD" and hit OK

11. Select "auto" installation and hit OK

12. Select every package except for "KDEI" and hit OK

13. Select "full" installation and hit OK

14. Insert the next Slackware installation disk (disk 2) when it prompts

you, and hit OK to continue

15. Now you will select the boot kernel that you will use from now on. If

you have a simple system with IDE drives you can probably just select "skip" and

go to the next step. If not then you should probably select "cdrom" and select

the kernel from the list that you selected to boot into the installation.

16. You can make a bootdisk if you like, but you don't have to.

17. For Modem you can select "no modem" and hit OK to continue

18. Enable hotplug, hit Yes to continue

19. Install lilo "simple" and hit OK to continue

20. lilo frame buffer console 640x480 is safe choice if you're not sure

21. Optional Lilo append, leave blank and hit OK to continue

22. Lilo destination, I usually choose MBR but root works most of the time

23. Mouse, select the mouse type that you have hooked up, or select ps/2

24. Load GPM at boot time, Hit Yes to continue

25. select Yes to configure your network

26. Hostname, we are typing "phone"

27. network, we are typing our local domain name

28. IP address, we are selecting Static IP, here's what we enter for

network, you should enter a network setup that will work with your local LAN:

- IP address: 10.10.10.15

- subnet: 255.255.0.0

- gateway: 10.10.10.1

- name server: 10.10.10.1

29. Accept your network settings

30. Startup services to run, change nothing and select OK to continue

31. select NO for custom screen fonts

32. Hardware clock to UTC, select NO

33. Select your time zone and hit OK

34. I usually select gnome as the window manager, even though you won't be

using it on this machine

35. Select Yes to enter a root password. type something that you will

remember.

36. Setup of Slackware Linux is complete, hit OK and EXIT then press

CTRL-ALT-DELETE to reboot your computer

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PHASE 2: COMPILING A CUSTOM LINUX KERNEL

From this step on you should be able to continue the installation remotely

although it is wise to at least have quick access to the machine if something

goes wrong.

To connect remotely through SSH on linux type "slogin serveripaddress" or to use

Windows to connect you can use a piece of free software called putty available

here: http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html

Also, for windows you can use SSH file transferring(SFTP) with a program called

filezilla: http://filezilla.sourceforge.net/

This is an optional step if your linux system is running, but compiling your own

custom kernel is always a way to optimize your system for the hardware you have

installed or a way to remove the unnecesary modules that are in the default

kernel. You will definately want to build your own kernel if you have a multi

processor machine. If you are new to Linux you probably do not want to do this.

If you are using a newer Digium Octasic-based echo-cancellation quad T1/E1 card

then you need to use a 2.6 Linux kernel in order to use the echo-cancellation

functionality of the card.

If you want to compile a 2.6 kernel then start with OPTION 1, otherwise to

compile the 2.4 kernel that comes with Slackware(2.4.33) start with OPTION 2:

OPTION 1: compile Linux kernel 2.6.17 *RECOMMENDED*

1. cd /usr/src

2. wget http://www.kernel.org/pub/linux/kernel/v2.6/linux-2.6.17.11.tar.gz

3. gunzip linux-2.6.17.11.tar.gz

4. tar xvf linux-2.6.17.11.tar

5. mv -f /usr/src/linux /usr/src/linux-old

6. ln -s /usr/src/linux-2.6.17.11 /usr/src/linux

7. cd linux

8. make mrproper # prep for kernel assembly

9. make menuconfig # launch configuration menu app

(this part is very dependant upon your own hardware)

(what is mentioned below are only changes beyond what is selected by default)

Block Layer --->

-> IO Scheduler

this should be set to CFQ if you do not have a hardware caching controller

if you have a hardware caching controller DEADLINE or NO-OP are the best

choices, if you have a battery backuped caching controller that is set to write-back you should use NO-OP.

Processor Type and Features --->

->Symmetric multi-processing support

(if you have multiple processors or a Dual-core or HT enabled)

->High Memory Support

(if you have more than 900MB of System RAM move upto 4GB)

->Timer frequency (1000 HZ)

(change to 1000Hz if using ztdummy for timer on pre 2.6.17 kernel)

->[*]Tickless System (enable UNLESS using ztdummy on pre 2.6.17 kernel)

->[*]Enable Kernel irq balancing

->Preemption Model (No Forced Preemption (Server))

(^This one is very important!!!)

Power management options (ACPI, APM) --->

->ACPI (Advanced Configuration and Power Interface) Support

(enable all down to Processor and thermal zone)

Bus options (PCI, PCMCIA, EISA, MCA, ISA) --->

->[*] PCI Express support

(if using Sangoma PCI Express card)

Networking --->

Amateur Radio support --->

<*> Amateur Radio AX.25 Level 2 protocol

[*] AX.25 DAMA Slave support

<*> Amateur Radio NET/ROM protocol

<*> Amateur Radio X.25 PLP (Rose)

(all needed for new Digium Octasic drivers)

Device Drivers --->

ATA/ATAPI/MFM/RLL support --->

<*> SCSI emulation support

(needed for SATA drives, also further down check chipset drivers)

SCSI device support --->

<*> RAID Transport Class

(needed if you are using a RAID)

SCSI low-level drivers --->

<*> Serial ATA (SATA) support

(required if using SATA drives)

(if using a SCSI RAID card pick correct driver here)

Multi-device support (RAID and LVM) --->

(select proper RAID types if using Linux RAID)

Network device support --->

Ethernet (10 or 100Mbit) --->

Ethernet (1000 Mbit) --->

(select proper drivers for the eype of network card you have)

Character devices --->

<*> Enhanced Real Time Clock Support

(double-check that this is enabled, very important)

Real Time Clock --->

<*> RTC class

(double-check that this is enabled, very important)

File systems --->

<*> Ext3 journalling file system support

(important if using ext3 filesystem)

pseudo filesystems --->

<*> Virtual memory file system support

(this is usually checked anyway but mandatory for recording to RAM[tmpfs])

Library routines --->

<*> CRC-CCITT functions

<*> CRC16 functions

<*> CRC32c (Castagnoli, et al) Cyclic Redundancy-Check

(important for new Digium Octasic drivers)

EXIT AND SAVE YOUR CONFIGURATION

10. make clean # clean up the kernel build areas

11. make bzImage # create a kernel bzImage

12. make modules # build the modules into the image

13. make modules_install # install kernel modules

14. cp arch/i386/boot/bzImage /boot/bzImage-XXXX # copy image

(put whatever you want in XXXX, that is your new kernel name)

15. cp System.map /boot/System.map-XXXX # copy system map

16. mv -f /boot/System.map /boot/System.map-orig

17. ln -s /boot/System.map-XXXX /boot/System.map # symlink map

18. vi /etc/lilo.conf # edit the lilo boot config file

image=/boot/bzImage-XXXX # add the new image in above-

label=test-XXXX # the previous one

root=/dev/hda1 # device of root partition

read-only

19. /sbin/lilo # run the lilo reload script

20. vi /etc/fstab # to include the following line for recording to RAM

tmpfs /var/spool/asterisk/monitor tmpfs rw 0 0

21. shutdown -r 0 # reboot machine and hope it worked

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OPTION 2: compile Linux kernel 2.4.33.3 (not recommended, very old)

1. cd /usr/src/linux # move to your linux source directory

2. cp .config config.save # copy old config to a save file

3. make mrproper # prep for kernel assembly

4. make menuconfig # launch configuration menu app

(this part is very dependant upon your own hardware)

enable processor version # select the processor that you have

enable SMP # if more than 1 processor or Intel HT

enable high memory () # if more than 1GB of RAM

enable SCSI Multiple # if SCSI drives

enable SCSI devices AMI Megaraid # if SCSI Megaraid adapter

enable 3com network devices # if 3com network card

enable ext3 file system # for ext3 to work

enable all ACPI options # for SMP to work

enable Enhanced Real Time Clock Support in Character devices section

# for SMP to work

enable any other hardware specific options

exit and save configuration

5. make dep # build the kernel dependancies

6. make clean # clean up the kernel build areas

7. make bzImage # create a kernel bzImage

8. make modules # build the modules into the image

9. make modules_install # install kernel modules

10. # nothing# mkinitrd /boot/initrd-XXXXXX.img XXXXXX *not needed on Slackware*

11. cp arch/i386/boot/bzImage /boot/bzImage-XXXXXX # copy image

(put whatever you want in XXXXXX, that is your new kernel name)

12. cp System.map /boot/System.map-XXXXXXN # copy system map

13. mv -f /boot/System.map /boot/System.map-orig

14. ln -s /boot/System.map-XXXXXX /boot/System.map # symlink map

15. vi /etc/lilo.conf # edit the lilo boot config file

image=/boot/bzImage-XXXXXX # add the new image in above-

label=test-XXXXXX # the previous one

root=/dev/hda1 # device of root partition

read-only

16. /sbin/lilo # run the lilo reload script

17. shutdown -r 0 # reboot machine and hope it worked

After compiling your kernel you can run a few commands to verify that you are

running your new kernel and that devices are running as they are supposed to:

ps --info (will show you your linux kernel version and other info)

cat /proc/cpuinfo (will show you processor type and more than one if SMP)

top (will show you system memory)

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PHASE 3: INSTALLING SOFTWARE BEFORE ASTERISK

In this step we will be installing software that Asterisk and/or astGUIclient

needs to be able to function to its fullest ability. Not all of these software

packages are manditory to successfully install Asterisk and some of them can be

installed on other machines on your network like MySQL or Apache. But, in this

installation we are assuming that there are no other machines on our network to

help the Asterisk server, so it must have everything it needs installed locally.

SUBPHASE 3.0: install new Gnu Make

A new version of the "make" compilation application replaced the 4-year-old

version that most Linux distros(Including Slackware) use. This is only needed

if you will be building Asterisk from the 1.4 release tree.

- wget http://mirrors.kernel.org/gnu/make/make-3.81.tar.gz

- gunzip make-3.81.tar.gz

- tar xvf make-3.81.tar

- cd make-3.81

- ./configure

- make

- make install

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SUBPHASE 3.1: MySQL requirements

You must at least have Mysql client installed on each VICIDIAL server, but you

only need one database server.

MySQL is a fast database system that is very easy to integrate with any

application. You can either install the server on the local Asterisk system or

have one somewhere on your network. For our purposes, we are creating an Asterisk

installation that is self contained and needs no other local servers to operate,

so we will need to install mysql on this machine.

*REQUIRED and OPTIONAL* (only install MySQL server locally if you don't want to use an installation

on another machine, Mysql client is required on all VICIDIAL servers)

NOTE: a minimum of MySQL server 4.0.X is required

You should increase the connect_timeout so connections do not fail on

a more loaded system.

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-5.0/mysql-5.0.67.tar.gz

- gunzip mysql-5.0.67.tar.gz

- tar xvf mysql-5.0.67.tar

- cd mysql-5.0.67

- groupadd mysql

- useradd -g mysql mysql

- ./configure --prefix=/usr/local/mysql --enable-shared=yes --with-readline --enable-thread-safe-client --enable-large-files --enable-assembler --with-client-ldflags=-all-static --with-mysqld-ldflags=-all-static --with-big-tables

**** If only MySQL client is needed for DBD::mysql then use this:

- "./configure --prefix=/usr/local/mysql --without-server --enable-shared=yes --with-readline"

- make

- make install

- PATH=$PATH:$HOME/bin:/usr/local/mysql/bin/

- export PATH

- PATH=$PATH:$HOME/bin:/usr/local/mysql/include/mysql/

- export PATH

- cp /usr/local/mysql-5.0.67/libmysql/.libs/libmysqlclient.so /usr/lib/

- cp /usr/local/mysql-5.0.67/libmysql/.libs/libmysqlclient.so.15 /usr/lib/

- cd /usr/local/mysql-5.0.67

- scripts/mysql_install_db

- chown -R root /usr/local/mysql

- chown -R mysql /usr/local/mysql/var

- chgrp -R mysql /usr/local/mysql

- cp support-files/my-huge.cnf /etc/my.cnf

- /usr/local/mysql/bin/mysqld_safe --user=mysql --skip-name-resolve --skip-host-cache &

- ln -s /tmp/mysql.sock /var/run/mysql/mysql.sock

- vi /etc/my.cnf

# add this line below 'skip-locking'

skip-name-resolve

# comment out the line 'log-bin=mysql-bin'

max_connections = 200

**** For some systems you may need to add the mysql/bin directory to your PATH:

- PATH=$PATH:$HOME/bin:/usr/local/mysql/bin/

- export PATH

**** you may also want to add those two lines to your /root/.bash_profile file

**** For Mysql 5 tree only, you also may need to copy the libmysqlclient.so file to libs

- cp /usr/local/mysql-5.0.67/libmysql/.libs/libmysqlclient.so /usr/lib/

- cp /usr/local/mysql-5.0.67/libmysql/.libs/libmysqlclient.so.15 /usr/lib/

- you are done

** INSTALLATION NOTE **

If you are having Linuxthreads problems upon onfigure, just execute the following command:

echo '/* Linuxthreads */' >> /usr/include/pthread.h

***** NOTE: if you will be using any of the DBI perl scripts: *****

Every machine that you will be using the newer BDI perl scripts on will need

to have the perl modules DBI and DBD::mysql installed on them. To do this

you will also need to at least have the MySQL client installed on the server

(see above) then you will need to go to 'cpan' and "install DBI" and

"install DBD::mysql". You may need to "force install DBD::mysql" if the DBD

tests fail on your first try, but that is OK since the tests are not needed

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SUBPHASE 3.2: Installing Perl Modules

NOTE - you can install ActiveState http://www.activestate.com perl which may

improve performance, but it is not required. Here's the source for ActiveState

Perl 5.8: (it's free)

http://downloads.activestate.com/ActivePerl/src/5.8/AP817_source.tgz

I hope to add the lengthy steps for installing it as your default perl on your

server but I need some time and a free machine to do that.

cpan is the "Comprehensive Perl Archive Network". It's a mirrored archive of

most of the perl modules out there complete with a installation and management

command-line interface. Here's what you do to start it:

*REQUIRED* (needed for perl AGIs)

1. perl -MCPAN -e shell # type in the command line

2. You will then go through CPAN setup, just hit ENTER for most prompts except

for the mirrors list, you will want to select at least 4 mirrors

- yes for manual configuration

- enter for the next 18 prompts

- for the "make install options" it's a good idea to add UNINST=1

- enter for the next 4 prompts

- select your continent and country

- select a few cpan mirrors

- enter for the next 2 prompts

3. Once you see the cpan> prompt you can begin installing modules

4. If you've never installed cpan before you should probably install the

following modules first: (say YES if asked to install prerequisites)

- install MD5

- install Digest::MD5

- install Digest::SHA1

- install readline (just hit Enter when it asks for operator)

- install Bundle::CPAN

- reload cpan

- then you can install other modules:

- install DBI

- force install DBD::mysql (must at least have mysqlclientlibs installed)

- install Net::Telnet

- install Time::HiRes

- install Net::Server

- install Switch

- install Mail::Sendmail

- install Unicode::Map (needed for super list loader Excel)

- install Jcode (needed for super list loader Excel)

- install Spreadsheet::WriteExcel (needed for super list loader Excel)

- install OLE::Storage_Lite (needed for super list loader Excel)

- install Proc::ProcessTable (needed for super list loader Excel)

- install IO::Scalar (needed for super list loader Excel)

- install Spreadsheet::ParseExcel (needed for super list loader Excel)

- if Spreadsheet::ParseExcel fails to install try running the following:

- force install Scalar::Util (this will enable weak references)

- install Spreadsheet::ParseExcel

- then quit cpan, you are done

5. Go to http://asterisk.gnuinter.net/ and download the asterisk-perl module

(backup link: http://download.vicidial.com/packages/asterisk-perl-0.08.tar.gz)

NOTE: Do NOT use the 0.09 version, it does not work with VICIDIAL

- to install this directly on the command line type:

- cd /usr/local

- wget http://asterisk.gnuinter.net/files/asterisk-perl-0.08.tar.gz

- gunzip asterisk-perl-0.08.tar.gz

- tar xvf asterisk-perl-0.08.tar

- cd asterisk-perl-0.08

- perl Makefile.PL

- make all

- make install

- you are done

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SUBPHASE 3.3: Installing other utilities

Sox is an audio utility that allows you to mix audio files together at their

start point into one file. it is necessary for Asterisk recordings that record

in and out as separate files

*REQUIRED* (needed for recording mixing)

1. Go to http://sourceforge.net/projects/sox/ and download the sox package

- to install this directly on the command line type:

- cd /usr/local

- wget http://easynews.dl.sourceforge.net/sourceforge/sox/sox-14.0.1.tar.gz

- gunzip sox-14.0.1.tar.gz

- tar xvf sox-14.0.1.tar

- cd sox-14.0.1

- ./configure --disable-shared

- make (if alsa.o errors add --disable-alsa-dsp to configure and redo)

- make install

- 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* (only needed if you will be converting recordings to MP3)

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://easynews.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

Screen is a terminal emulator that allows you to run a process as command line

and be able to detach from them('Ctrl+a' then 'd') and log all output of the

terminal to a screenlog file if desired(add a '-L' to the launching command).

In our installations this is how we launch Asterisk upon startup and still have

the ability to log output and still attach to the screen that executed asterisk

originally.

*REQUIRED* *MANDITORY FOR VICIDIAL SERVERS*

3. 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

or for older version:

- wget http://mirrors.kernel.org/gnu/screen/screen-3.9.15.tar.gz

- gunzip screen-4.0.2.tar.gz

- tar xvf screen-4.0.2.tar

- cd screen-4.0.2

- ./configure

- make

- make install

- you are done

ttyload is a simple terminal application that shows the processor load in a

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

package

- to install this directly on the command line type:

- cd /usr/local

- wget http://www.daveltd.com/src/util/ttyload/ttyload-0.4.4.tar.gz

- gunzip ttyload-0.4.4.tar.gz

- tar xvf ttyload-0.4.4.tar

- cd ttyload-0.4.4

- make

- ln -s /usr/local/ttyload-0.4.4/ttyload /usr/bin/ttyload

- you are done

ntpd is the network time protocol daemon that matches the time on your machine

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 SERVERS* (install on server and all clients)

5. 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-4.2/ntp-4.2.2p3.tar.gz

If you get compilation errors here try 4.1.2:

- wget http://www.eecis.udel.edu/~ntp/ntp_spool/ntp4/ntp-4.1.2.tar.gz

- gunzip ntp-4.2.2p3.tar.gz

- tar xvf ntp-4.2.2p3.tar

- cd ntp-4.2.2p3

- ./configure

- make

- make install

- vi /etc/ntp.conf (change to just 1 line: "server ntp.myfloridacity.us")

- cp /etc/ntp.conf /etc/ntpd.conf # just to be sure

- /usr/local/bin/ntpdate -u ntp.myfloridacity.us # initial sync

- /usr/sbin/ntpd # run it

- you are done

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*

NOTE: another good network analysis utility is "iptraf" and is on most system

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6. 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-0.9.4.tar.gz

- gunzip libpcap-0.9.4.tar.gz

- tar xvf libpcap-0.9.4.tar

- cd libpcap-0.9.4

- ./configure

- make

- make install

- cd /usr/local

- wget http://www.ex-parrot.com/~pdw/iftop/download/iftop-0.17.tar.gz

- gunzip iftop-0.17.tar.gz

- tar xvf iftop-0.17.tar

- cd iftop-0.17

- ./configure

- make

- make install

- iftop

- you are done

ploticus is a free graph creation package that allows you to create line graphs

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* (only needed for server performance graphing web reports)

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7. 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

- wget http://superb-west.dl.sourceforge.net/sourceforge/ploticus/pl240src.tar.gz

- gunzip pl240src.tar.gz

- tar xvf pl240src.tar

- cd pl240src/src/

- make clean

- make

- make install

- you are done

NOTE: uncomment these lines to compile on systems without X11(v232):

NOXFLAG = -DNOX11

XLIBS =

XOBJ =

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

openssh is a remote login protocol server that is always a good idea to have

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]

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8. 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

- gunzip zlib-1.2.3.tar.gz

- tar xvf zlib-1.2.3.tar

- cd zlib-1.2.3

- ./configure

- make

- make install

- cd /usr/local

- wget http://ftp.arcane-networks.fr/pub/OpenBSD/OpenSSH/portable/openssh-5.2p1.tar.gz

- gunzip openssh-5.2p1.tar.gz

- tar xvf openssh-5.2p1.tar

- cd openssh-5.2p1

- ./configure

- make

- make install

- you are done

openssl is the open-source SSL libraries package, and to install a fake SSL cert

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)

9. 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-0.9.8j.tar.gz

- gunzip openssl-0.9.8j.tar.gz

- tar xvf openssl-0.9.8j.tar

- cd openssl-0.9.8j

- ./config

- make

- make install

- you are done

apache is a web server that allows you to use many different modules with it to

extend it's functionality. In order to use some of the astguiclient

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)

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10. 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

- wget http://mirror.nyi.net/apache/httpd/httpd-2.2.11.tar.gz

- gunzip httpd-2.2.11.tar.gz

- tar xvf httpd-2.2.11.tar

- wget http://us2.php.net/distributions/php-5.2.9.tar.gz

- gunzip php-5.2.9.tar.gz

- tar xvf php-5.2.9.tar

- cd httpd-2.2.11

- ./configure --enable-so --with-apxs2

- make

- make install

- cd ../php-5.2.9

- ./configure --with-apxs2=/usr/local/apache2/bin/apxs --with-mysql

- make

- make install

- cp php.ini-dist /usr/local/lib/php.ini

NOTE: you will want to make sure NOTICE logging is turned off:

error_reporting = E_ALL & ~E_NOTICE ; (this is default)

!!! REQUIRED !!! be sure the memory limit for scripts in php.ini is AT LEAST 48M:

memory_limit = 48M

Make sure short tags are enabled:

short_open_tag = On

some other fields to change if using web-based lead loader:

max_execution_time = 330

max_input_time = 360

post_max_size = 48M

upload_max_filesize = 42M

default_socket_timeout = 360

- vi /usr/local/apache2/conf/httpd.conf

add the following lines:

"AddType application/x-httpd-php .php .phtml"

"LoadModule php4_module libexec/libphp5.so"

"LoadModule php4_module modules/libphp5.so"

modify the index.html line and add index.php to the list

to disable logging, change:

"CustomLog logs/access_log common"

to this:

"CustomLog /dev/null common"

to enable web browsing of Recordings on Asterisk server, add this:

Alias /RECORDINGS/ "/var/spool/asterisk/monitorDONE/"

Options Indexes MultiViews

AllowOverride None

Order allow,deny

Allow from all

Forcetype application/forcedownload

- /usr/local/apache2/bin/apachectl start

- go to http://your-new-asterisk-server-ipaddress/ to see if it worked

- you are done

NOTE: If using PHP5 you may need to add the following line to php.ini:

short_open_tag = On

OPTIONAL- Load eAccelerator PHP-caching application:

Even so this is technically an optional part, it is strongly recommended that you install eAccelerator

since it will slash PHPs processing power requirements greatly. Without eAccelerator load on the system

can be ten times as high, which can cause all kinds of problems, this is especially true for single

system setups.

- Go to http://eaccelerator.net and download the most recent package

- cd /usr/local

- wget http://bart.eaccelerator.net/source/0.9.5/eaccelerator-0.9.5.3.zip

- unzip eaccelerator-0.9.5.3.zip

- cd eaccelerator-0.9.5.3

- export PHP_PREFIX="/usr/local"

- $PHP_PREFIX/bin/phpize

- ./configure --enable-eaccelerator=shared --with-php-config=$PHP_PREFIX/bin/php-config

- make

- make install

- vi /usr/local/lib/php.ini

Add the following lines to the dynamic extensions section of php.ini:

(you may need to change the extension location depending on your install of php)

zend_extension="../../../usr/local/eaccelerator-0.9.5/modules/eaccelerator.so"

eaccelerator.shm_size="48"

eaccelerator.cache_dir="/tmp/eaccelerator"

eaccelerator.enable="1"

eaccelerator.optimizer="1"

eaccelerator.check_mtime="1"

eaccelerator.debug="0"

eaccelerator.filter=""

eaccelerator.shm_max="0"

eaccelerator.shm_ttl="0"

eaccelerator.shm_prune_period="0"

eaccelerator.shm_only="0"

eaccelerator.compress="1"

eaccelerator.compress_level="9"

- mkdir /tmp/eaccelerator

- chmod 0777 /tmp/eaccelerator

# to verify installation:

- php -v

balance is a load-balancing application for Linux that will allow you to spread

the load of your web traffic across many servers. If you are running more than

70 agents on a single server you may want to install this application and build

another cheap web server to handle the extra load.

*OPTIONAL*

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11. 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

- gunzip balance-3.34.tgz

- tar xvf balance-3.34.tar

- cd balance-3.34

- make

- make install

- /usr/sbin/balance -f 81 localhost:80 10.10.10.16:80

That command will take port 81 traffic and send it evenly to the local

server and the 10.10.10.16 server reducing the load and speeding up

the applications. More info on balance: http://www.inlab.de/balance.pdf

subversion is the new code control framework use by the Asterisk community. If

you want to use the latest development code of Asterisk you will need to have

this loaded on your system.

*OPTIONAL*

12. 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-1.5.2.tar.gz

- gunzip subversion-1.5.2.tar.gz

- tar xvf subversion-1.5.2.tar

- cd subversion-1.5.2

- ./configure

- make

- make install

mtop is a great utility for real-time monitoring of mysql and the queries that

are running in it.

*OPTIONAL*

13. Go to http://mtop.sourceforge.net to download the most recent version

- to install this directly on the command line type:

- cd /usr/local

- wget http://superb-east.dl.sourceforge.net/sourceforge/mtop/mtop-0.6.6.tar.gz

- gunzip mtop-0.6.6.tar.gz

- tar xvf mtop-0.6.6.tar

- cd mtop-0.6.6

- cpan

- install Curses

- install Getopt::Long

- install Net::Domain

- quit

- perl Makefile.PL

- make

- make install

- /usr/local/bin/mtop --dbuser=root --seconds=3

sipsak is an optional utility that VICIDIAL can use to send messages to an

agent's SIP-based phone(like the Snom 320) to display text on their LCD screen.

If you want to use this, make sure it is installed on the same server that your

web server is installed on(Apache).

*OPTIONAL*

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14. Go to http://sipsak.org to download the most recent version

- to install this directly on the command line, type:

- cd /usr/local

- wget http://download.berlios.de/sipsak/sipsak-0.9.6-1.tar.gz

- gunzip sipsak-0.9.6-1.tar.gz

- tar xvf sipsak-0.9.6-1.tar

- cd sipsak-0.9.6-1

- ./configure

- make

- make install

- /usr/local/bin/sipsak --version

PHASE 4: INSTALLING ASTERISK

OK, all the prep work is done, now it's time to start having fun with Asterisk.

There are two basic ways to install Asterisk, an official release(at the time of

this writing the official release is 1.2.30.2) and the SVN_DEV version(development

branch). We recommend using Asterisk 1.2.30.2. 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.

NOTE: If you want to use Asterisk 1.4, you will need to make sure that you set

the servers table "asterisk_version" field to the proper version number and you

can use the docs/conf_examples/extensions.conf.sample-1.4 file for your default

dialplan

NOTE: If you want to use release 1.0.8 or 9 we would recommend either using the

CVS_v1-0 branch where the issues are fixed, or patching your 1.0.8/1.0.9 code

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 '1.0.11.1'

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://downloads.digium.com/pub/asterisk/releases/asterisk-1.2.30.2.tar.gz

- wget http://downloads.digium.com/pub/zaptel/releases/zaptel-1.2.27.tar.gz

- wget http://downloads.digium.com/pub/libpri/releases/libpri-1.2.5.tar.gz

- gunzip asterisk-1.2.30.2.tar.gz

- tar xvf asterisk-1.2.30.2.tar

- gunzip zaptel-1.2.27.tar.gz

- tar xvf zaptel-1.2.27.tar

- gunzip libpri-1.2.5.tar.gz

- tar xvf libpri-1.2.5.tar

B. if you want latest SVN_1.2 version (release tree with new patches)

- svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2

- svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2

- svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2

C. if you want latest SVN_DEV version (not recommended) [1.6 tree]

- svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk

- svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel

- svn checkout http://svn.digium.com/svn/libpri/trunk libpri

- ALL ->

- (1.0 tree)if you want to allow for more than 100 voicemail messages in a

mailbox(warning this will slightly increase memory usage when a call

is in voicemail) edit the voicemail source code file:

- vi /usr/src/asterisk/asterisk/apps/app_voicemail.c

edit this line and change 100 to 999:

#define MAXMSG 100

- (1.0 tree)if you have no X server installed on your Asterisk machine, then you

will need to comment out the gtk console lib(only affects 1.0 releases)

edit the voicemail source code file:

- vi /usr/src/asterisk/asterisk/pbx/Makefile

edit this line at the top and just add a hash # in front of it as shown:

#PBX_LIBS+=$(shell $(CROSS_COMPILE_BIN)gtk-config --cflags >/dev/null 2>/dev/null && echo "pbx_gtkconsole.so")

- cd ./zaptel-1.2.27

- make clean

- make

- make install

- cd ../libpri-1.2.5

- make clean

- make

- make install

- cd ../asterisk-1.2.30.2

- (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

- cd apps

- wget http://www.eflo.net/files/app_amd2.c

- mv app_amd2.c app_amd.c

- vi Makefile

replace this line(line 32):

app_mixmonitor.so app_stack.so

with this line:

app_mixmonitor.so app_stack.so app_amd.so

- wget http://www.eflo.net/files/amd2.conf

- mkdir /etc/asterisk

- mv amd2.conf /etc/asterisk/amd.conf

*OPTIONAL*(1.2.23 thru 1.2.30.2) apply the meetme DTMF passthru patch

- wget http://www.eflo.net/files/meetme_DTMF_passthru-1.2.23.patch

- patch -p1 < ./meetme_DTMF_passthru-1.2.23.patch

- File to patch: app_meetme.c

*OPTIONAL*(1.2.12.1 thru 1.2.30.2) apply the meetme volume control patch

*Different patches available for 1.2.7.1 through 1.2.14

- wget http://www.eflo.net/files/meetme_volume_control_1.2.16.patch

- patch -p1 < ./meetme_volume_control_1.2.16.patch

- File to patch: app_meetme.c

- cd ../

-(1.2 tree) apply the cli delimiter patch

- wget http://www.eflo.net/files/cli_chan_concise_delimiter.patch

- patch -p1 < ./cli_chan_concise_delimiter.patch

- File to patch: cli.c

-(gcc version 4.2) apply the gsm audio codec patch to fix gsm

- wget http://download.vicidial.com/asterisk-patches/1.2-gsm-gcc4.2.patch

- patch -p1 ./codecs/gsm/Makefile 1.2-gsm-gcc4.2.patch

*OPTIONAL*(1.2.14 thru 1.2.30.2) rewrite of waitforsilence

- wget http://download.vicidial.com/asterisk-patches/app_waitforsilence.c

- mv -f app_waitforsilence.c apps/app_waitforsilence.c

*OPTIONAL* shorter enter and leave sounds for meetme

- wget http://www.eflo.net/files/enter.h

- wget http://www.eflo.net/files/leave.h

- mv -f enter.h apps/enter.h

- mv -f leave.h apps/leave.h

- make clean

- make

- make install

- make samples # this makes sample conf files (only use for new installs)

- modprobe zaptel # this loads the zaptel module

- install the module for the digium device that you are using, we are

using the T100P single span T1 card so we use:

- modprobe wct1xxp

Here's the list of all digium cards and the modules you use with

them:

Card Module

-----------------

TDM400P wctdm

X100P wcfxo

TDM* wcfxs

S100U wcusb

T100P wct1xxp

E100P wct1xxp

T400P tor2

E400P tor2

TE110P wcte11xp

TE410P wct4xxp

TE405P wct4xxp

TE411P wct4xxp

TE406P wct4xxp

TE210P wct2xxp

TE205P wct2xxp

TDM2400P wctdm24xxp

- If you have chosen a Sangoma T1/E1 or analog card, you will need to

follow their instructions for installation of their driver software

LATEST Sangoma Wanpipe drivers:

ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-3.3.9.tgz

- now your asterisk installation is built and loaded and it's time to

configure it.

NOTES: If you want to install zttool diagnostics you may need the newt package installed:

- wget http://download.vicidial.com/packages/newt-0.51.6.tar.gz

- gunzip newt-0.51.6.tar.gz

- tar xvf newt-0.51.6.tar

- cd newt-0.51.6

- ./configure

- make

- make install

- cd ../

- ln -s /usr/lib/libnewt.so.0.51.6 /usr/lib/libnewt.so.0.51

then go to your zaptel folder and do 'make zttool'

Digium/Clone X100P EXAMPLE:

Here is an example of a configuration where an X100P single FXO card is used for

zaptel timing and not used for calling:

NOTE: you can get an X100P through ebay for $10-$30 that will work with Asterisk

/etc/zaptel.conf:

loadzone=us

defaultzone=us

fxsks=1

/etc/asterisk/zapata.conf:

[trunkgroups]

[channels]

context=unused

signalling=fxs_ks

channel => 1

Added this to the rc.local file:

# Load zaptel drivers for x100p

modprobe zaptel

modprobe wcfxo

If you will be doing native music-on-hold for your inbound calls, you will need

musiconhold audio files to be converted to native formats like GSM, ULAW and ALAW:

cd /var/lib/asterisk/mohmp3/

mpg123 -s --rate 44100 --mono /var/lib/asterisk/mohmp3/fpm-sunshine.mp3 > /var/lib/asterisk/mohmp3/fpm-sunshine.raw

sox -r 44100 -w -s -c 1 fpm-sunshine.raw -r 8000 -c 1 fpm-sunshine.wav

sox fpm-sunshine.wav -t gsm -r 8000 -b -c 1 fpm-sunshine.gsm

sox fpm-sunshine.wav -t ul -r 8000 -b -c 1 fpm-sunshine.pcm

mpg123 -s --rate 44100 --mono /var/lib/asterisk/mohmp3/fpm-calm-river.mp3 > /var/lib/asterisk/mohmp3/fpm-calm-river.raw

sox -r 44100 -w -s -c 1 fpm-calm-river.raw -r 8000 -c 1 fpm-calm-river.wav

sox fpm-calm-river.wav -t gsm -r 8000 -b -c 1 fpm-calm-river.gsm

sox fpm-calm-river.wav -t ul -r 8000 -b -c 1 fpm-calm-river.pcm

mpg123 -s --rate 44100 --mono /var/lib/asterisk/mohmp3/fpm-world-mix.mp3 > /var/lib/asterisk/mohmp3/fpm-world-mix.raw

sox -r 44100 -w -s -c 1 fpm-world-mix.raw -r 8000 -c 1 fpm-world-mix.wav

sox fpm-world-mix.wav -t gsm -r 8000 -b -c 1 fpm-world-mix.gsm

sox fpm-world-mix.wav -t ul -r 8000 -b -c 1 fpm-world-mix.pcm

mkdir ../orig-mp3

mv -f *.mp3 ../orig-mp3/

mkdir ../quiet-mp3

cd ../quiet-mp3

sox -r 44100 -w -s -c 1 ../mohmp3/fpm-sunshine.raw -r 8000 -c 1 fpm-sunshine.wav vol 0.25

sox fpm-sunshine.wav -t gsm -r 8000 -b -c 1 fpm-sunshine.gsm

sox fpm-sunshine.wav -t ul -r 8000 -b -c 1 fpm-sunshine.pcm

sox -r 44100 -w -s -c 1 ../mohmp3/fpm-calm-river.raw -r 8000 -c 1 fpm-calm-river.wav vol 0.25

sox fpm-calm-river.wav -t gsm -r 8000 -b -c 1 fpm-calm-river.gsm

sox fpm-calm-river.wav -t ul -r 8000 -b -c 1 fpm-calm-river.pcm

sox -r 44100 -w -s -c 1 ../mohmp3/fpm-world-mix.raw -r 8000 -c 1 fpm-world-mix.wav vol 0.25

sox fpm-world-mix.wav -t gsm -r 8000 -b -c 1 fpm-world-mix.gsm

sox fpm-world-mix.wav -t ul -r 8000 -b -c 1 fpm-world-mix.pcm

rm -f ../mohmp3/*.raw

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PHASE 5: CONFIGURING ASTERISK AND YOUR SIP PHONES

As of release 2.0.5 it is now possible to configure SIP and IAX phones and

carrier trunks without editing conf files, just by using the web administration

interface. For more information on this, please read the VICIDIAL Manager

Manual available at www.eflo.net

In this phase we will configure the telco lines, the SIP phones, the extensions,

meetme(conference calling) rooms, dialplan extensions and the voicemail boxes.

After this phase your Asterisk system should be able to place and receive calls

to and from the SIP phones you have installed over the telco lines you've hooked

up. There are several things that we will not be showing how to do because

Asterisk is extremely flexible and has so many different ways of being

configured, that if we were to try to explain them all in this document it would

be 99% asterisk configuration and be 20,000 lines long, and that would

just be a barrier for those who just want to get it set up. The "Wiki" and the

mailing list are two very good resources for finding answers if you run into

problems configuring your system, here are links to them:

The Wiki: http://www.voip-info.org/tiki-index.php

The Lists: http://www.asterisk.org/index.php?menu=support

I need to note that it is possible to install Asterisk and use astGUIclient

applications with no Zaptel(Digium/Sangoma/Rhino/etc...) cards installed, but it

is not recommended even if you are not going to use Zap trunks for your inbound-

outbound calls with no real Zap devices, you would need to use a dummy timer

(zt_dummy) based on you USB ports to get meetme conference rooms working

properly and you may have other issues along the way. We would at least

recommend getting a X100 or X101 board from Digium or a clone manufacturer so

there is a dedicated hardware timer in place on your system.

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SUBPHASE 5.0: setting up your Asterisk configuration files

1. edit zaptel.conf

- vi /etc/zaptel.conf

There are many examples inside of the zaptel.conf file that is

generated with the "make samples" command that we issued at the end

of the last phase. There are many different parameters for the

different telco line possibilities, because we are installing a T1

that is NON-PRI-isdn B8ZS ExtendedSuperframe(ESF) E&M Wink start and 24

channels, we will use the following settings for zaptel.conf:

span=1,1,0,esf,b8zs

e&m=1-24

loadzone = us

defaultzone=us

FOR A PRI YOU WOULD USE SOMETHING LIKE THIS:

span=2,2,0,esf,b8zs

bchan=25-47

dchan=48

2. edit zapata.conf

- vi /etc/asterisk/zapata.conf

There are also many examples of how to configure zapata.conf online.

we decided to separate our T1 into two line groups to keep some

incoming calls from being busy if we filled up all of our lines.

Here's what we used(you can set echocancel=no if you are using PRIs):

[channels]

group=1

language=en

signalling=em_w

usecallerid=yes

callerid=asreceived

context=default

echocancel=64

echocancelwhenbridged=yes

rxgain=1.0

txgain=1.0

channel => 1-2

group=2

language=en

signalling=em_w

usecallerid=yes

callerid=asreceived

context=default

echocancel=64

echocancelwhenbridged=yes

rxgain=1.0

txgain=1.0

channel => 3-24

FOR A PRI YOU WOULD USE SOMETHING LIKE THIS:

group=3

language=en

signalling=pri_net

usecallerid=yes

callerid=asreceived

callprogress=no

busydetect=no

context=default

echocancel=64

echocancelwhenbridged=yes

rxgain=1.0

txgain=1.0

channel => 25-47

3. edit sip.conf

As of release 2.0.5 it is now possible to configure SIP and IAX phones and

carrier trunks without editing conf files, just by using the web administration

interface. For more information on this, please read the VICIDIAL Manager

Manual available at www.eflo.net

- vi /etc/asterisk/sip.conf

here is where we will edit the configuration of our SIP compatible

phone devices. As stated at the beginning, we will be setting up a

Grandstream Budgetone 102 phone and a Sipura/Linksys SPA-2000 adapter with

two analog phones connected(each with it's own extension). Here are

the settings we used to set each of them up:

[general]

port = 5060

bindaddr = 0.0.0.0

context = default

; register SIP account on remote machine if using SIP trunks

; register => testSIPtrunk:test@10.10.10.16:5060

;

; setup account for SIP trunking:

; [SIPtrunk]

; disallow=all

; allow=ulaw

; allow=alaw

; type=friend

; username=testSIPtrunk

; secret=test

; host=10.10.10.16

; dtmfmode=inband

; qualify=1000

[gs102]

disallow=all

allow=ulaw

allow=alaw

type=friend

username=gs102

secret=test

host=dynamic

dtmfmode=inband

defaultip=10.10.10.16

qualify=1000

mailbox=102

[spa2000]

disallow=all

allow=ulaw

allow=alaw

type=friend

username=spa2000

secret=test

host=dynamic

dtmfmode=inband

defaultip=10.10.10.17

qualify=1000

mailbox=2000

[spa2001]

disallow=all

allow=ulaw

allow=alaw

type=friend

username=spa2001

secret=test

host=dynamic

dtmfmode=inband

defaultip=10.10.10.17

qualify=1000

mailbox=2001

4. edit meetme.conf

- vi /etc/asterisk/meetme.conf

This is known as the conference calling configuration file. We are

just going to add two conferences(one without a pin number and one

with a pin number required for entry):

[rooms]

conf => 8600

conf => 8601,1234

5. edit iax.conf

As of release 2.0.5 it is now possible to configure SIP and IAX phones and

carrier trunks without editing conf files, just by using the web administration

interface. For more information on this, please read the VICIDIAL Manager

Manual available at www.eflo.net

- vi /etc/asterisk/iax.conf

This is the IAX configuration file, below is a very simple config for

having two Asterisk servers connect natively to each other, if you

will be using this, make sure to add the optional lines included

after the extensions.conf section. Also, there is an account setup

here for a firefly IAX softphone to use.(details on that later)

* IMPORTANT NOTE * if you plan to use IAX2 trunks for VICIDIAL

outbound dialing you must register with the remote IAX2 server

through the iax.conf file, not just in the Dial or TRUNK line

of the extensions.conf dialplan.

[general]

bindport=4569

iaxcompat=yes

bandwidth=high

allow=all

allow=gsm ; Always allow GSM, it's cool :)

jitterbuffer=no

tos=lowdelay

[ASTtest2]

type=friend

accountcode=IAXtrunk2

context=default

auth=plaintext

host=dynamic

permit=0.0.0.0/0.0.0.0

secret=test

disallow=all

allow=ulaw

qualify=yes

[firefly01]

type=friend

accountcode=firefly01

context=default

auth=plaintext

host=dynamic

permit=0.0.0.0/0.0.0.0

secret=test

disallow=all

allow=gsm

qualify=yes

##### EXAMPLE - This is a config example for setting up Binfone service(http://www.binfone.com)

NOTE: The "1112223333" is your iax username. When you signup, Binfone creates

a default IAX username for you, (a 5 digit number, usually, starting with a 1).

This works for most customers. Folks that want inbound then also sign up for

DIDs, each of which has its own IAX username. (Which is the 10 digit DID).

Each username has its own password which is managed through their web interface.

NOTE: If you will be using the G729 codec through binfone there is now a

dedicated G729 gateway that only handles G729 calls. Please use this address to

iax.conf:

[general]

[1112223333]

auth=md5

type=friend

username=1112223333

secret=PASSWORD

host=iax.binfone.com

context=incoming-IAX-context-in-extensions.conf

extensions.conf:

[global]

TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface

[default]

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN:1},55,o)

exten => _91NXXNXXXXXX,3,Hangup

[incoming]

exten => 1112223333,1,AGI(agi://127.0.0.1:4577/call_log)

exten => 1112223333,2,Dial(sip/gs102,55,o)

exten => 1112223333,3,Hangup

dnsmgr.conf: # It is very helpful to enable dnsmgr

[general]

enable=yes ; enable creation of managed DNS lookups

refreshinterval=300 ; refresh managed DNS lookups every seconds

##### END EXAMPLE

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6. edit voicemail.conf

- vi /etc/asterisk/voicemail.conf

This is where we set up the voicemail boxes for the extensions that

we have set up:

[general]

format=wav49|gsm|wav

serveremail=asterisk

attach=yes

skipms=3000

maxsilence=10

silencethreshold=128

maxlogins=3

[zonemessages]

eastern=America/New_York|'vm-received' Q 'digits/at' IMp

central=America/Chicago|'vm-received' Q 'digits/at' IMp

central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M 'hours'

[default]

102 => 102,Grandstream Mailbox,root@localhost

2000 => 2000,Sipura Mailbox 1

2001 => 2001,Sipura Mailbox 2

3001 => 3001,Firefly Mailbox 1

7. edit manager.conf

- vi /etc/asterisk/manager.conf

This is where we set up remote logins to the asterisk manager

interface, to allow sending of Action commands from remote

connections to the Asterisk server, this will be important for the

astguiclient applications so let's set that up now:

[general]

enabled = yes

port = 5038

bindaddr = 0.0.0.0

[cron]

secret = 1234

read = system,call,log,verbose,command,agent,user

write = system,call,log,verbose,command,agent,user

[updatecron]

secret = 1234

read = command

write = command

[listencron]

secret = 1234

read = system,call,log,verbose,command,agent,user

write = command

[sendcron]

secret = 1234

read = command

write = system,call,log,verbose,command,agent,user

8. edit logger.conf

- vi /etc/asterisk/logger.conf

This file determines the messages that are logged to the console and

the /var/log/asterisk/messages file. We usually turn on full logging

to the messages file to more easily diagnose any problems that we may

run into, the problem with this is that is does produce very large

files, so be warned:

[logfiles]

console => notice,warning,error

messages => notice,warning,error,debug,verbose

9. edit extensions.conf

- vi /etc/asterisk/extensions.conf

You should be using the sample extensions.conf that is included with the

release of VICIDIAL that you installed, below is just an explanation of what

most of those entries do and why they are there.

This is known as the dialplan. Since we are installing a

Long-Distance T1 with one 800 number on it, we will need to put that

800 number in the plan, as well as how to dial out through the T1

lines and we will need to add an entry for each of the phones that we

have just set up in the sip.conf file. There are many examples both

in the sample file and online for what to put in your dialplan, here

is the simplified dialplan that we are using:

######------ START extensions.conf example ------######

[general]

static=yes

writeprotect=no

[globals]

CONSOLE=Console/dsp ; Console interface for demo

TRUNK=Zap/g1 ; Trunk interface

TRUNKX=Zap/g2 ; 2nd trunk interface

TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface

TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind monitoring

TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing

[default]

; Extension 8600 + 8601 conference rooms

exten => 8600,1,Meetme,8600

exten => 8601,1,Meetme,8601

; Extension 102 - Grandstream hardphone

exten => 102,1,Playback,transfer|skip ; "Please hold while..."

exten => 102,2,Dial,sip/gs102|20|to ; Ring, 20 secs max

exten => 102,3,Voicemail,u102 ; Send to voicemail...

; Extension 2000 Sipura line 1

exten => 2000,1,Dial,sip/spa2000|30|to ; Ring, 30 secs max

exten => 2000,2,Voicemail,u2000 ; Send to voicemail...

; Extension 2001 Sipura line 2

exten => 2001,1,Dial,sip/spa2001|30|to ; Ring, 30 secs max

exten => 2001,2,Voicemail,u2001 ; Send to voicemail...

; Extension 2020 rings both sipura lines

exten => 2001,1,Dial,sip/spa2000&sip/spa2001|30|to ; Ring, 30 secs max

exten => 2001,2,Voicemail,u2000 ; Send to voicemail...

; Extension 301 rings the firefly softphone

exten => 301,1,Dial,(IAX2/firefly01@firefly01/s)

exten => 301,2,Hangup

; Extension 3429 - Inbound 800 number (1-800-555-3429)

exten => _**3429,1,Ringing

exten => _**3429,2,Answer

exten => _**3429,3,Dial,sip/spa2000&sip/spa2001|30|to ; Ring, 30 secs max

exten => _**3429,4,Voicemail,u2000 ; Send to voicemail...

; Extension 3429 - with ANI [callerID]

exten => _*NXXNXXXXXX*3429,1,Ringing

exten => _*NXXNXXXXXX*3429,2,Answer

exten => _*NXXNXXXXXX*3429,3,Dial,sip/spa2000&sip/spa2001|30|to ; Ring, 30

secs max

exten => _*NXXNXXXXXX*3429,4,Voicemail,u2000 ; Send to voicemail...

; dial a long distance outbound number to the UK

exten => _901144XXXXXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},55,tTo)

exten => _901144XXXXXXXXXX,2,Hangup

; dial a long distance outbound number to Australia

exten => _901161XXXXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},,tTo)

exten => _901161XXXXXXXXX,2,Hangup

; dial an 800 outbound number

exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)

exten => _91800NXXXXXX,2,Hangup

exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)

exten => _91888NXXXXXX,2,Hangup

exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)

exten => _91877NXXXXXX,2,Hangup

exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)

exten => _91866NXXXXXX,2,Hangup

; dial a local 727 outbound number with area code

exten => _9727NXXXXXX,1,Dial(${TRUNK}/1${EXTEN:1},,tTo)

exten => _9727NXXXXXX,2,Hangup

; dial a local 813 outbound number with area code

exten => _9813NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,tTo)

exten => _9813NXXXXXX,2,Hangup

; dial a long distance outbound number

exten => _91NXXNXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},,tTo)

exten => _91NXXNXXXXXX,2,Hangup

; dial a local outbound number (modified because of only LD T1)

exten => _9NXXXXXX,1,Dial(${TRUNK}/1727${EXTEN:1},,tTo)

exten => _9NXXXXXX,2,Hangup

; barge monitoring extension

exten => 8159,1,ZapBarge

exten => 8159,2,Hangup

; # timeout invalid rules

exten => #,1,Playback(invalid) ; "Thanks for trying the demo"

exten => #,2,Hangup ; Hang them up.

exten => t,1,Goto(#,1) ; If they take too long, give up

exten => i,1,Playback(invalid) ; "That's not valid, try again"

; Give voicemail at extension 8500

exten => 8500,1,VoicemailMain

exten => 8500,2,Goto(s,6)

; ASTERISK AGENTS LOGINS FOR QUEUES (NOT part of VICIDIAL)

; the following assumes phone agent login and exten are 3 digits and the same

; also assumes that 3-digit login is present in agents.conf and queueus.conf

;Agent Logout then stay onhook, DIAL 54 + 3-digit ID

exten => _54XXX,1,AgentCallbackLogin(||)

; the following are used to login and logout of Asterisk Queues from phone

;Agent Login then stay offhook on the phone, DIAL 55 + 3-digit ID

exten => _55XXX,1,AgentLogin(${EXTEN:1})

;Agent Login then stay onhook, phones will ring, DIAL 56 + 3-digit ID

exten => _56XXX,1,AgentCallbackLogin(||${EXTEN:1}@default)

######------ END extensions.conf example ------######

### OPTIONAL IAX trunk extensions entries for long distance dialing over IAX

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN:1},55,o)

exten => _91NXXNXXXXXX,3,Hangup

### OPTIONAL SIP trunk extensions entries for long distance dialing over SIP

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@SIPtrunk,55,o)

exten => _91NXXNXXXXXX,3,Hangup

### OPTIONAL IAX Load Balance extens to allow for Overflow and Balanced VDAD

### In this setup, the serverIP is the prefix followed by agent conf_exten

### FOR MORE INFORMATION, READ THE LOAD_BALANCING.txt DOCUMENT

### server 1 extens:

exten => _010*010*010*016*.,1,Dial(${TRUNKIAX2}/${EXTEN:16},55,o)

### server 2 extens:

exten => _010*010*010*015*.,1,Dial(${TRUNKIAX1}/${EXTEN:16},55,o)

10. edit dnsmgr.conf: # It is very helpful to enable dnsmgr

[general]

enable=yes ; enable creation of managed DNS lookups

refreshinterval=300 ; refresh managed DNS lookups every seconds

11. Now that you have configured Asterisk, it is time to try to start it for the

first time.

- First make sure that your T1 line(or other telco line) is connected to the

digium card.

- next type the following at the command prompt: "ztcfg -vvvvvv"

- you should see a confirmation that the Zaptel device has loaded

- now you can launch asterisk with the following command:

"asterisk -vvvvvvvvvvvvgc"

- you should see a lot of messages scroll by and at the end you should

be given a CLI> prompt if everything loaded OK. To get out of Asterisk you can

type "stop now". Now that you are sure it is running you can either run it in a

separate terminal window or use the start_asterisk_boot.pl that you will

install with astguiclient to start Asterisk:

/usr/share/astguiclient/start_asterisk_boot.pl

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SUBPHASE 5.1: setting up your SIP phones

You will need to follow the instructions for the phones that you are using with

your system, but here's the way to set up a Grandstream 102 and a Linksys/Sipura

SPA-2000

1. Here are basic instructions for setting up a Grandstream BT 102:

- On the phone plug it in to power only at first and follow these

instructions:

- wait for the phone to boot up and press the MENU button

- go through the menu screens with the menu key and the up/down arrow

keys to move from setting to setting. We are setting the following

values:

- DHCP OFF

- IP Addr: 010.010.010.016

- Subnet: 255.255.000.000

- router: 010.010.010.001

- dnS: 010.010.010.001

- tftp: we leave this blank for now

- menu 7 we change the codec to G-711u

- now wait 10 seconds and unplug the power and plug it back in

- you can also plug the network cable into the LAN port on the phone

- at this time you can go to your workstation and open a new web browser

- go to http://10.10.10.16/ the password is "admin"

- here is where you will enter in the configuration details for the

phone to register with the Asterisk server

- SIP server: 10.10.10.15

- SIP user ID: gs102

- Authenticate ID: gs102

- Password: test

- Name: gs102

- Voice Mail UserID: 102

- Send DTMF: in-audio

- NTP Server: tick.mit.edu

- then click update, click review changes, and click reboot

- your phone should now be able to register with the Asterisk server. If you

still have your console screen up you should see a registration message

appear telling you that gs102 has registered.

2. Here are the basic instructions for setting up a Sipura SPA-2000 analog

adapter with 2 lines.

- Plug power and two analog phones into the adapter.

- pick up the phone plugged into line1 and press **** to enter admin menu

- press 101# then 0# to disable DHCP

- press 111# then 10*10*10*17# to change the IP address

- press 121# then 255*255*0*0# to change the subnet mask

- press 131# then 10*10*10*1# to change the default gateway

- hang up the phone, unplug the power, plug in the network cable and plug in

the power cable

- now you can go the the admin website: http://10.10.10.17/admin/advanced

- you will need to make these setting changes:

- click on the "Line 1" tab at the top and change the following values:

- Proxy: 10.10.10.15

- Display Name: spa2000

- userID: spa2000

- password: test

- authID: spa2000

- change the dialplan to the following:

- then click the "submit all changes" button at the bottom of the page

and your first phone line should work now

- to register the second line, simply click on the "Line 2" tab and go

through the above steps except use spa2001 instead of spa2000 for the use IDs

3. Now both of your phone devices are set up and you can try making phone calls

between the three phones

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SUBPHASE 5.2: setting up an IAX2 phone

This is optinal and we won't go into too much detail about this, but currently

there are several IAX hard and softphones on the market and more are coming

every month. Follow the instructions with the IAX phone you have chosen and

follow the steps below:

1. Add an entry into your iax.conf file like below if you have not already

[firefly01]

type=friend

accountcode=firefly01

context=default

auth=plaintext

host=dynamic

permit=0.0.0.0/0.0.0.0

secret=test

qualify=yes

2. Add an entry into your extensions.conf file like below if it is not in there

; Extension 3001 rings IAX phone

exten => 301,1,Dial(IAX2/firefly01@firefly01/s)

exten => 301,2,Voicemail,u301 ; Send to voicemail...

3. Download Firefly 3rd party, or IDEfisk for Windows or Linux:

- http://www.virbiage.com/firefly/download/firefly-thirdparty.exe

MIRRORS:

- http://mirror.isp.net.au/ftp/pub/firefly/firefly-thirdparty.exe

- http://download.vicidial.com/softphones/firefly-thirdparty.exe

IDEFISK:

- http://www.asteriskguru.com/idefisk/

- Install the application

- Launch Firefly Softphone

- click the "I wish to connect to a 3rd party network" button

- Enter in network name: Asterisk

- Select IAX2 as the protocol

- enter in your server address: "10.10.10.15" in our case

- enter login and pass: "firefly01" and "test" for in our case

- click OK and you should be logged in and can place calls

SUBPHASE 5.3: setting up a Zap phone

This is optinal and we won't go into too much detail about this either, there

are a few ways to use Zap devices as phones on your Asterisk system: Zaptel

phone cards, Channel Banks going through Zaptel T1 card, outside line call

coming in going through Zaptel line card. There isn't much to do but set your

Zaptel config files up and put entries into your extensions.conf file:

1. Add an entry into your extensions.conf file like below

; Extension 4001 rings Zap phone

exten => 4001,1,Dial,Zap/1|30| ; ring Zap device 1

exten => 4001,2,Voicemail,u4001 ; Send to voicemail...

PHASE 6: INSTALLING ASTGUICLIENT AND VICIDIAL

Now that Asterisk is installed and running we can add the astGUIclient and

VICIDIAL components to the system.

SUBPHASE 6.0: putting the files in place

There are two methods for downloading astGUIclient/VICIDIAL, a release and SVN

1. Go to http://astguiclient.sf.net/ and download the latest astguiclient

package(as of this writing it is 2.0.5)

- for 2.0.X release:

- mkdir /usr/src/astguiclient

- cd /usr/src/astguiclient

- wget http://internap.dl.sourceforge.net/sourceforge/astguiclient/astguiclient_2.0.5.zip

- unzip astguiclient_2.0.5.zip

- perl install.pl (make sure you are in the directory with the install.pl file)

- for SVN 2.0.5 branch:

- mkdir /usr/src/astguiclient

- cd /usr/src/astguiclient

- svn checkout svn://svn.eflo.net:43690/agc_2-X/branches/agc_2.0.5

- cd agc_2.0.5

- perl install.pl

- for SVN 2.2 trunk:

- mkdir /usr/src/astguiclient

- cd /usr/src/astguiclient

- svn checkout svn://svn.eflo.net:43690/agc_2-X/trunk

- cd trunk

- perl install.pl

select to do interactive setup and customize to your server

NOTE: if this is a fresh install, it is strongly suggested that you

select 'Y' to copy the sample conf files.

- there is one more file you need that's not included with the download

package, it's the conf.gsm file(this is the half-hour music file that we use

to put people on hold). I have a free classical music file that is available

free for download at the following two sites:

http://download.vicidial.com/sounds/conf.gsm

http://astguiclient.sf.net/conf.gsm

Once you have downloaded it, you will need to copy it to this folder:

/var/lib/asterisk/sounds/

Then you will need to execute this command to copy it as the park file

'cp /var/lib/asterisk/sounds/conf.gsm /var/lib/asterisk/sounds/park.gsm'

Here are the steps spelled out:

cd /var/lib/asterisk/sounds

wget http://download.vicidial.com/sounds/conf.gsm

cp conf.gsm park.gsm

- you are done

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SUBPHASE 6.1: creating the MySQL "asterisk" database

we will create the database and add a few initial records so that we can

use the administrative web interface. Since this is a new install it is easier

to use our new mysql script file to add the tables to the database:

1. at the command prompt type go to the mysql client:

/usr/local/mysql/bin/mysql

2. type the following into the mysql client prompt:

(make sure you put your IP address in place of "10.10.10.15" in the queries below)

######------ BEGIN Mysql data entry(you can copy and paste this into terminal) #

create database asterisk;

NOTE: if you will be using lead files with a language that does not use the

standard latin character set then you will want to use UTF8 for your default

characterset in the MySQL database. This requires at least MySQL 4.1.11 and you

can use the following query to create the database:

CREATE DATABASE `asterisk` DEFAULT CHARACTER SET utf8 COLLATE utf8_unicode_ci;

GRANT SELECT,INSERT,UPDATE,DELETE,LOCK TABLES on asterisk.* TO cron@'%' IDENTIFIED BY '1234';

GRANT SELECT,INSERT,UPDATE,DELETE,LOCK TABLES on asterisk.* TO cron@localhost IDENTIFIED BY '1234';

GRANT RELOAD ON *.* TO cron@'%';

GRANT RELOAD ON *.* TO cron@localhost;

flush privileges;

# NOTE: if using MySQL 4.1.X or higher(not 5.X) you may need to run this query too:

UPDATE mysql.user set password=OLD_PASSWORD('1234') where user='cron';

# To make sure that new processes can connect to the Database under load we should

# increase the global connect_timeout

SET GLOBAL connect_timeout=60;

# NOTE: make sure you do NOT put any spaces or other punctuation in the

# server_id, phone, extension, or user fields in the queries below if you edit them.

use asterisk;

\. /usr/src/astguiclient/trunk/extras/MySQL_AST_CREATE_tables.sql

or you may need to run this if you get an error:

\. /usr/src/astguiclient/agc_2.0.5/extras/MySQL_AST_CREATE_tables.sql

\. /usr/src/astguiclient/astguiclient/MySQL_AST_CREATE_tables.sql

### to load in default IAX and SIP phone accounts run the following query

\. /usr/src/astguiclient/trunk/extras/sip-iax_phones.sql

or you may need to run this if you get an error:

\. /usr/src/astguiclient/agc_2.0.5/extras/sip-iax_phones.sql

\. /usr/src/astguiclient/astguiclient/sip-iax_phones.sql

### to load the initial server values for this first system install

\. /usr/src/astguiclient/trunk/extras/first_server_install.sql

or you may need to run this if you get an error:

\. /usr/src/astguiclient/agc_2.0.5/extras/first_server_install.sql

\. /usr/src/astguiclient/astguiclient/first_server_install.sql

quit

to populate the timezone/country table run this command from command line:

- /usr/share/astguiclient/ADMIN_area_code_populate.pl

to load the performance testing leads run these commands:

- cp /usr/src/astguiclient/trunk/extras/performance_test_leads.txt /usr/share/astguiclient/LEADS_IN/

- cp /usr/src/astguiclient/agc_2.0.5/extras/performance_test_leads.txt /usr/share/astguiclient/LEADS_IN/

- cp /usr/src/astguiclient_2.0.5/trunk/extras/performance_test_leads.txt /usr/share/astguiclient/LEADS_IN/

- /usr/share/astguiclient/VICIDIAL_IN_new_leads_file.pl --forcelistid=107 --forcephonecode=1

######------ END Mysql data entry ------######

NOTE: If you will be using channelbanks for agent phones you can use the

/extras/single_channelbank_phones.sql file to help enter the phones table entries.

NOTE: If you will be using IAX or SIP phones for agent phones you can use the

/extras/sip-iax_phones.sql file to help enter the phones table entries.

NOTE: if you are not installing using default user/pass or have MySQL on another

server, you will need to edit either the /etc/astguiclient.conf file or the

dbconnect.php files in the astguiclient, vicidial and agc directories of your

webroot.

3. Enter the vicidial administration page:

http://10.10.10.15/vicidial/admin.php

NOTE: if you click on the Logout button you must leave the user/pass empty and click OK

- Here you will enter the login and password that you inserted into the mysql

database in the vicidial_users table (subphase 6.1 [6666/1234])

- Now that you are logged into the astGUIclient administration system we can

add a new phone entry for each of the sipura lines we created.

- click on the "Admin" link at the top, then the "ADD PHONE" link below

that and enter in the proper information for each of the new phone lines.

Here's what we entered for spa2000:

- Phone extension: spa2000

- Dialplan Number: 2000

- Voicemail Box: 2000

- Phone IP address: 10.10.10.17

- Computer IP address: 10.10.9.17

- Server IP: 10.10.10.15

- Login: spa2000

- Password: test

- Status: ACTIVE

- Active Account: Y

- Phone Type: Sipura SPA-2000 line 1

- Full Name: Sipura line 1 test

- Company: TEST

- Picture:

- for the next phone simply replace 2000 with 2001 in the above example

- now your phones are all all set up in the astguiclient system and you can

use this website to add new phones to be used with astguiclient and monitor

the number of calls people are making.

- now your database is set up for the astguiclient conferences which

will allow you to have over 6 remote parties that you called from your

GUI client application in one conference.

- click on the "LIST ALL SERVERS" link at the top then click on the

server to modify. Verify that the GMT time zone and all other fields

are what you want them to be. There is a setting(Max VICIDIAL Trunks)

that can be modified to limit the number of VICIDIAL outbound trunks

that will be allowed to use on this server.

4. **OPTIONAL** For IAX clients you will need to use full phone name as the

extension on the admin page entry: "firefly01@firefly01" for our IAX phone example

previously. And do not forget to set the protocol on this page to IAX2

5. **OPTIONAL** For Zap clients you will need to use full Zap Channel name as the

extension on the admin page entry: "1-1" for our Zap phone example

previously. And do not forget to set the protocol on this page to Zap

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SUBPHASE 6.2: making additions to your Asterisk conf files

Now that the database is set up and our phones have entries in the system we can

make the additions to the running Asterisk system that will allow astguiclient

to work with it.

Again, if you have selected to use the sample conf files during installation

then you do not have to add any of these lines to your conf files, they should

already be included.

1. Add the call_log entries to all incoming/outgoing extensions entries:

- here is how our sample dialplan changes for adding call_log entries(only

effected extension groups are show):

######------ START extensions.conf changes for call_log ------######

##### This 'h' exten is VERY important for VICIDIAL usage,

##### you will have problems if it is not in your dialplan!

exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; MANDITORY VDAD extens:

; In this setup, the serverIP is the prefix followed by agent conf_exten

; These lines are REQUIRED for VICIDIAL to work properly

; local server extens:

; BE SURE TO CHANGE THIS LINE FOR YOUR IP ADDRESS!

exten => _010*010*010*015*.,1,Goto(default,${EXTEN:16},1)

exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)

exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)

; Local blind monitoring

exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)

; OPTIONAL server 2 extens, needed for load balancing:

exten => _010*010*010*016*.,1,Dial(${TRUNKIAX2}/${EXTEN:16},55,o)

; Extension 3429 - Inbound 800 number (1-800-555-3429)

exten => _**3429,1,Ringing

exten => _**3429,2,AGI(agi://127.0.0.1:4577/call_log)

exten => _**3429,3,Answer

exten => _**3429,4,Dial,sip/spa2000&sip/spa2001,30,to

exten => _**3429,5,Voicemail,u2000

; Extension 3429 - with ANI [callerID]

exten => _*NXXNXXXXXX*3429,1,Ringing

exten => _*NXXNXXXXXX*3429,2,AGI(agi://127.0.0.1:4577/call_log)

exten => _*NXXNXXXXXX*3429,3,Answer

exten => _*NXXNXXXXXX*3429,4,Dial,sip/spa2000&sip/spa2001,30,to

exten => _*NXXNXXXXXX*3429,5,Voicemail,u2000

; Extension 7275551212 - Inbound local number from PRI with 10 digit delivery

; SEE extensions.conf.sample-1.4 for changes needed for use with Asterisk 1.4

exten => 7275551212,1,Ringing

exten => 7275551212,2,Wait(1)

exten => 7275551212,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID}-----${CALLERIDNUM}-----${CALLERIDNAME})

exten => 7275551212,4,Answer

exten => 7275551212,5,Dial,sip/spa2000&sip/spa2001,30,to

exten => 7275551212,6,Voicemail,u2000

; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,

; dial a long distance outbound number to the UK

exten => _901144XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _901144XXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},55,tTo)

exten => _901144XXXXXXXXXX,3,Hangup

; dial a long distance outbound number to Australia

exten => _901161XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _901161XXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,tTo)

exten => _901161XXXXXXXXX,3,Hangup

; Extensions for performance testing

exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _91999NXXXXXX,2,Dial(${TRUNKloop}/${EXTEN:2},,tTo)

exten => _91999NXXXXXX,3,Hangup

exten => 999999999999,1,AGI(agi://127.0.0.1:4577/call_log)

exten => 999999999999,2,Dial(${TRUNKloop}/${EXTEN:1},,tTo)

exten => 999999999999,3,Hangup

; dial an 800 outbound number

exten => _91800NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _91800NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTo)

exten => _91800NXXXXXX,3,Hangup

exten => _91888NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _91888NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTo)

exten => _91888NXXXXXX,3,Hangup

exten => _91877NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _91877NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTo)

exten => _91877NXXXXXX,3,Hangup

exten => _91866NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _91866NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTo)

exten => _91866NXXXXXX,3,Hangup

; dial a local 727 outbound number with area code

exten => _9727NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _9727NXXXXXX,2,Dial(${TRUNK}/1${EXTEN:1},,tTo)

exten => _9727NXXXXXX,3,Hangup

; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,

; dial a long distance outbound number

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,tTo)

exten => _91NXXNXXXXXX,3,Hangup

; dial a local outbound number (modified because of only LD T1)

exten => _9NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

exten => _9NXXXXXX,2,Dial(${TRUNK}/1727${EXTEN:1},,tTo)

exten => _9NXXXXXX,3,Hangup

######------ END extensions.conf changes ------######

2. Add the call_inbound entries to all incoming extensions entries that you want

CallerID popups on:

- here is how our sample dialplan changes for adding call_inbound

entries(only effected extension groups are show):

######------ START extensions.conf changes for call_inbound ------######

; parameters for call_inbound.agi (7 fields separated by five dashes "-----"):

; 1. the extension of the phone to ring as defined in the asterisk.phones table

; 2. the phone number that was called, for the live_inbound/_log entry

; 3. a text description of the number that was called in

; 4-7. optional fields, they are also passed as fields in the GUI to web browser

; Extension 3429 - Inbound 800 number (1-800-555-3429)

exten => _**3429,1,Ringing

exten => _**3429,2,AGI(agi://127.0.0.1:4577/call_log)

exten => _**3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)

exten => _**3429,4,Answer

exten => _**3429,5,Dial,sip/spa2000&sip/spa2001|30|to

exten => _**3429,6,Voicemail,u2000

; Extension 3429 - with ANI [callerID]

exten => _*NXXNXXXXXX*3429,1,Ringing

exten => _*NXXNXXXXXX*3429,2,AGI(agi://127.0.0.1:4577/call_log)

exten => _*NXXNXXXXXX*3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)

exten => _*NXXNXXXXXX*3429,3,Answer

exten => _*NXXNXXXXXX*3429,4,Dial,sip/spa2000&sip/spa2001|30|to

exten => _*NXXNXXXXXX*3429,5,Voicemail,u2000

; parameters for agi-VDAD_ALL_inbound.agi (9 fields separated by five dashes "-----"):

; 1. the method of call handling for the script:

; - CID - CID received, add record with phone number

; - CIDLOOKUP - Lookup CID to find record in whole system

; - CIDLOOKUPRL - Restrict lookup to one list

; - CIDLOOKUPRC - Restrict lookup to one campaign's lists

; - CLOSER - Closer calls from VICIDIAL fronters

; - ANI - ANI received, add record with phone number

; - ANILOOKUP - Lookup ANI to find record in whole system

; - ANILOOKUPRL - Restrict lookup to one list

; - 3DIGITID - Enter 3 digit code to go to agent

; - 4DIGITID - Enter 4 digit code to go to agent

; - 5DIGITID - Enter 5 digit code to go to agent

; - 10DIGITID - Enter 10 digit code to go to agent

; 2. the method of searching for an available agent:

; - LO - Load Balance Overflow only (priority to home server)

; - LB - Load Balance total system

; - SO - Home server only

; 3. the full name of the IN GROUP to be used in vicidial for the inbound call

; 4. the phone number that was called, for the log entry

; 5. the callerID or lead_id of the person that called(usually overridden)

; 6. the park extension audio file name if used

; 7. the status of the call initially(usually not used)

; 8. the list_id to insert the new lead under if it is new (and CID/ANI available)

; 9. the phone dialing code to insert with the new lead if new (and CID/ANI available)

; 10. the campaign_id to search within lists if CIDLOOKUPRC

; inbound VICIDIAL call with CID delivery through T1 PRI

exten => 1234,1,Answer ; Answer the line

exten => 1234,2,AGI(agi-VDAD_ALL_inbound.agi,CID-----LB-----CL_GALLERIA-----7274515134-----Closer-----park----------999-----1)

exten => 1234,3,Hangup

; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover, IAX or SIP channel]

exten => _90009.,1,Answer ; Answer the line

exten => _90009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)

exten => _90009.,3,Hangup

exten => _990009.,1,Answer ; Answer the line

exten => _990009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)

exten => _990009.,3,Hangup

### follow these instructions if you plan to have VICIDIAL agents take inbound or closer calls:

1. in VICIDIAL web admin "add a new in-group" (the above examples would be "CL_GALLERIA")

- group IDs cannot contain spaces ' ' or dashes '-' or plusses '+'

- if you are using a HEX color value make sure to include the hash '#' at the beginning

2. create a new campaign in VICIDIAL called "CLOSER" and set "allow inbound blended" to Y

3. check the CL_GALLERIA checknox in the "Allowed In-Groups" section

4. have agents log in to the CLOSER campaign and select the CL_GALLERIA in-group

5. they will now start receiving inbound calls

6. as calls come in, each call is inserted into the vicidial_list table under the

list specified int the AGI string, In the above example that would be list 999

7. if you want to take closer calls from the campaign "TEST" you will need to create

an in-group called "CL_TEST_" for internal closing(on the same system) or "CL_TEST_L"

for local closing(closer on different system from fronter) and then the fronter will

click on the "internal closer" button to send the call to a closer

* NOTE, you need to set the dial_level of the CLOSER campaign to 1 or higher for inbound/closers to work

######------ END extensions.conf changes for call_inbound ------######

3. Add the ZapBarge entries for all zap lines:

- here is how our sample dialplan changes for adding zapbarge line-specific

entries(this is a pure addition, nothing is being modified):

; ZapBarge direct channel extensions

exten => _86120XX,1,ZapBarge(${EXTEN:5})

4. Add the meetme entries for astguiclient and VICIDIAL conferences to

meetme.conf:

- here is how our sample meetme.conf file changes for adding conference

entries (this is a pure addition, nothing is being modified):

######------ START meetme.conf additions for conferences ------######

conf => 8600001

conf => 8600002

conf => 8600003

conf => 8600004

conf => 8600005

conf => 8600006

conf => 8600007

conf => 8600008

conf => 8600009

conf => 8600010

conf => 8600011

conf => 8600012

conf => 8600013

conf => 8600014

conf => 8600015

conf => 8600016

conf => 8600017

conf => 8600018

conf => 8600019

conf => 8600020

conf => 8600021

conf => 8600022

conf => 8600023

conf => 8600024

conf => 8600025

conf => 8600026

conf => 8600027

conf => 8600028

conf => 8600029

conf => 8600030

conf => 8600031

conf => 8600032

conf => 8600033

conf => 8600034

conf => 8600035

conf => 8600036

conf => 8600037

conf => 8600038

conf => 8600039

conf => 8600040

conf => 8600041

conf => 8600042

conf => 8600043

conf => 8600044

conf => 8600045

conf => 8600046

conf => 8600047

conf => 8600048

conf => 8600049

conf => 8600050

conf => 8600051

conf => 8600052

conf => 8600053

conf => 8600054

conf => 8600055

conf => 8600056

conf => 8600057

conf => 8600058

conf => 8600059

conf => 8600060

conf => 8600061

conf => 8600062

conf => 8600063

conf => 8600064

conf => 8600065

conf => 8600066

conf => 8600067

conf => 8600068

conf => 8600069

conf => 8600070

conf => 8600071

conf => 8600072

conf => 8600073

conf => 8600074

conf => 8600075

conf => 8600076

conf => 8600077

conf => 8600078

conf => 8600079

conf => 8600080

conf => 8600081

conf => 8600082

conf => 8600083

conf => 8600084

conf => 8600085

conf => 8600086

conf => 8600087

conf => 8600088

conf => 8600089

conf => 8600090

conf => 8600091

conf => 8600092

conf => 8600093

conf => 8600094

conf => 8600095

conf => 8600096

conf => 8600097

conf => 8600098

conf => 8600099

conf => 8600100

conf => 8600101

conf => 8600102

conf => 8600103

conf => 8600104

conf => 8600105

conf => 8600106

conf => 8600107

conf => 8600108

conf => 8600109

conf => 8600110

conf => 8600111

conf => 8600112

conf => 8600113

conf => 8600114

conf => 8600115

conf => 8600116

conf => 8600117

conf => 8600118

conf => 8600119

conf => 8600120

conf => 8600121

conf => 8600122

conf => 8600123

conf => 8600124

conf => 8600125

conf => 8600126

conf => 8600127

conf => 8600128

conf => 8600129

conf => 8600130

conf => 8600131

conf => 8600132

conf => 8600133

conf => 8600134

conf => 8600135

conf => 8600136

conf => 8600137

conf => 8600138

conf => 8600139

conf => 8600140

conf => 8600141

conf => 8600142

conf => 8600143

conf => 8600144

conf => 8600145

conf => 8600146

conf => 8600147

conf => 8600148

conf => 8600149

conf => 8600150

conf => 8600151

conf => 8600152

conf => 8600153

conf => 8600154

conf => 8600155

conf => 8600156

conf => 8600157

conf => 8600158

conf => 8600159

conf => 8600160

conf => 8600161

conf => 8600162

conf => 8600163

conf => 8600164

conf => 8600165

conf => 8600166

conf => 8600167

conf => 8600168

conf => 8600169

conf => 8600170

conf => 8600171

conf => 8600172

conf => 8600173

conf => 8600174

conf => 8600175

conf => 8600176

conf => 8600177

conf => 8600178

conf => 8600179

conf => 8600180

conf => 8600181

conf => 8600182

conf => 8600183

conf => 8600184

conf => 8600185

conf => 8600186

conf => 8600187

conf => 8600188

conf => 8600189

conf => 8600190

conf => 8600191

conf => 8600192

conf => 8600193

conf => 8600194

conf => 8600195

conf => 8600196

conf => 8600197

conf => 8600198

conf => 8600199

conf => 8600200

######------ END meetme.conf additions for conferences ------######

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5. Add the conference entries for astguiclient conferences:

- here is how our sample dialplan changes for adding conference entries (this

is a pure addition, nothing is being modified):

; astGUIclient conferences

exten => _86000[0-4]X,1,Meetme,${EXTEN}|q

6. Add the conference entries for VICIDIAL conferences:

- here is how our sample dialplan changes for adding VICIDIAL conference

entries(this is a pure addition, nothing is being modified):

NOTE: see below these entries for app_conference instructions is used

######------ START extensions.conf changes for VD conf ------######

exten => _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})

exten => _X48600XXX,2,Hangup

exten => _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})

exten => _X38600XXX,2,Hangup

exten => _X28600XXX,1,MeetMeAdmin(${EXTEN:2},m,${EXTEN:0:1})

exten => _X28600XXX,2,Hangup

exten => _X18600XXX,1,MeetMeAdmin(${EXTEN:2},M,${EXTEN:0:1})

exten => _X18600XXX,2,Hangup

exten => _55558600XXX,1,MeetMeAdmin(${EXTEN:4},K)

exten => _55558600XXX,2,Hangup

exten => 8300,1,Hangup

; VICIDIAL conferences

exten => _86000[5-9]X,1,Meetme,${EXTEN}|F

exten => _86001XX,1,Meetme,${EXTEN}|F

exten => _8600200,1,Meetme,${EXTEN}|F

; quiet entry and leaving conferences for VICIDIAL

exten => _78600XXX,1,Meetme,${EXTEN:1}|Fq

; quiet monitor extensions for meetme rooms (for room managers)

exten => _68600XXX,1,Meetme,${EXTEN:1}|Fmq

; Local blind monitoring

exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)

; voicelab exten

exten => _86009XX,1,Meetme,${EXTEN}|Fmq

; voicelab exten moderator

exten => _986009XX,1,Meetme,${EXTEN:1}

######------ END extensions.conf changes for VD conf ------######

NOTE: If you want to do DTMF passthru with app_conference bee sure to add the

"i" and "t" flags to the 8600XX lines: Conference(8600051|it)

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7. Add the more entries for astGUIclient specific uses:

- here are some more dialplan additions needed to use astGUIclient(this is a

pure addition, nothing is being modified):

######------ START extensions.conf other additions ------######

; park channel for client GUI parking, hangup after 30 minutes

; create a GSM formatted audio file named "park.gsm" that is 30 minutes long

; and put it in /var/lib/asterisk/sounds

exten => 8301,1,Answer

exten => 8301,2,AGI(park_CID.agi)

exten => 8301,3,Playback(park)

exten => 8301,4,Hangup

exten => 8303,1,Answer

exten => 8303,2,AGI(park_CID.agi)

exten => 8303,3,Playback(conf)

exten => 8303,4,Hangup

; park channel for client GUI conferencing, hangup after 30 minutes

; create a GSM formatted audio file named "conf.gsm" that is 30 minutes long

; and put it in /var/lib/asterisk/sounds

exten => 8302,1,Answer

exten => 8302,2,Playback(conf)

exten => 8302,3,Hangup

exten => 8304,1,Answer

exten => 8304,2,Playback(ding)

exten => 8304,3,Hangup

; default audio for safe harbor 2-second-after-hello message then hangup

; create a GSM formatted audio file complies with safe harbor rules

; and put it in /var/lib/asterisk/sounds then change filename below

exten => 8307,1,Answer

exten => 8307,2,Playback(vm-goodbye)

exten => 8307,3,Hangup

; this is used for recording conference calls, the client app sends the filename

; value as a callerID recordings go to /var/spool/asterisk/monitor (WAV)

; SEE extensions.conf.sample-1.4 for changes needed for use with Asterisk 1.4

exten => 8309,1,Answer

exten => 8309,2,Monitor(wav,${CALLERIDNAME})

exten => 8309,3,Wait,3600

exten => 8309,4,Hangup

; this is used for recording conference calls, the client app sends the filename

; value as a callerID recordings go to /var/spool/asterisk/monitor (GSM)

; SEE extensions.conf.sample-1.4 for changes needed for use with Asterisk 1.4

exten => 8310,1,Answer

exten => 8310,2,Monitor(gsm,${CALLERIDNAME})

exten => 8310,3,Wait,3600

exten => 8310,4,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL

; replace conf with the message file you want to leave

exten => 8320,1,WaitForSilence(2000,2) ; AMD got machine. leave message after recording

exten => 8320,2,Playback(conf)

exten => 8320,3,AGI(VD_amd_post.agi,${EXTEN})

exten => 8320,4,Hangup

; this is used to allow the GUI to send you directly into voicemail

; don't forget to set GUI variable $voicemail_exten to this extension

; SEE extensions.conf.sample-1.4 for changes needed for use with Asterisk 1.4

exten => 8501,1,VoicemailMain(s${CALLERIDNUM})

exten => 8501,2,Hangup

; this is used to allow the GUI to send live calls directly into voicemail

; don't forget to set GUI variable $voicemail_dump_exten to this extension

exten => _85026666666666.,1,Wait(2)

exten => _85026666666666.,2,Voicemail(${EXTEN:14})

exten => _85026666666666.,3,Hangup

; this is used for sending DTMF signals within conference calls, the client app

; sends the digits to be played in the callerID field

; sound files must be placed in /var/lib/asterisk/sounds

exten => 8500998,1,Answer

exten => 8500998,2,Playback(silence)

exten => 8500998,3,AGI(agi-dtmf.agi)

exten => 8500998,4,Hangup

; prompts for recording AGI script, ID is 4321

; first variable is format (gsm/wav)

; second variable is timeout in milliseconds (default is 360000 [6 minutes])

exten => 8167,1,Answer

exten => 8167,2,AGI(agi-record_prompts.agi,wav-----360000)

exten => 8167,3,Hangup

exten => 8168,1,Answer

exten => 8168,2,AGI(agi-record_prompts.agi,gsm-----360000)

exten => 8168,3,Hangup

; playback of recorded prompts

exten => _851XXXXX,1,Answer

exten => _851XXXXX,2,Playback(${EXTEN})

exten => _851XXXXX,3,Hangup

#### VDAD STANDARD TRANSFER ENTRIES ####

; VICIDIAL_auto_dialer transfer script for no-agent campaigns:

exten => 8364,1,AGI(agi://127.0.0.1:4577/call_log)

exten => 8364,2,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

exten => 8364,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

exten => 8364,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

exten => 8364,5,Hangup

; VICIDIAL_auto_dialer transfer script:

exten => 8365,1,AGI(agi://127.0.0.1:4577/call_log)

exten => 8365,2,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)

exten => 8365,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)

exten => 8365,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)

exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:

exten => 8366,1,AGI(agi://127.0.0.1:4577/call_log)

exten => 8366,2,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)

exten => 8366,3,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)

exten => 8366,4,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)

exten => 8366,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:

exten => 8367,1,AGI(agi://127.0.0.1:4577/call_log)

exten => 8367,2,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)

exten => 8367,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)

exten => 8367,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)

exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:

exten => 8368,1,AGI(agi://127.0.0.1:4577/call_log)

exten => 8368,2,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

exten => 8368,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

exten => 8368,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

exten => 8368,5,Hangup

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:

exten => 8369,1,AGI(agi://127.0.0.1:4577/call_log)

exten => 8369,2,AMD(2000|2000|1000|5000|120|50|4|256)

exten => 8369,3,AGI(VD_amd.agi,${EXTEN})

exten => 8369,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

exten => 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

exten => 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

exten => 8369,7,Hangup

; VICIDIAL auto-dial reminder script

exten => 8372,1,AGI(agi://127.0.0.1:4577/call_log)

exten => 8372,2,AGI(agi-VDADautoREMINDER.agi,${EXTEN})

exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})

exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})

exten => 8372,5,Hangup

; VICIDIAL SURVEY transfer script AMD with Load Balanced:

exten => 8373,1,AGI(agi://127.0.0.1:4577/call_log)

exten => 8373,2,AMD(2000|2000|1000|5000|120|50|4|256)

exten => 8373,3,AGI(VD_amd.agi,${EXTEN})

exten => 8373,4,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)

exten => 8373,5,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)

exten => 8373,6,Hangup

#### VDAD SIP UNREGISTERED TRANSFER ENTRIES ####

#### Use these entries IN PLACE OF the entries above if you are using SIP trunks

#### and are not registering your provider in sip.conf

;; VICIDIAL_auto_dialer transfer script for no-agent campaigns:

;exten => 8364,1,Playback(sip-silence)

;exten => 8364,2,AGI(agi://127.0.0.1:4577/call_log)

;exten => 8364,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

;exten => 8364,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

;exten => 8364,5,Hangup

;

;; VICIDIAL_auto_dialer transfer script:

;exten => 8365,1,Playback(sip-silence)

;exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)

;exten => 8365,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)

;exten => 8365,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)

;exten => 8365,5,Hangup

;

;; VICIDIAL_auto_dialer transfer script SURVEY at beginning:

;exten => 8366,1,Playback(sip-silence)

;exten => 8366,2,AGI(agi://127.0.0.1:4577/call_log)

;exten => 8366,3,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)

;exten => 8366,4,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)

;exten => 8366,5,Hangup

;

;; VICIDIAL_auto_dialer transfer script Load Balance Overflow:

;exten => 8367,1,Playback(sip-silence)

;exten => 8367,2,AGI(agi://127.0.0.1:4577/call_log)

;exten => 8367,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)

;exten => 8367,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)

;exten => 8367,5,Hangup

;

;; VICIDIAL_auto_dialer transfer script Load Balanced:

;exten => 8368,1,Playback(sip-silence)

;exten => 8368,2,AGI(agi://127.0.0.1:4577/call_log)

;exten => 8368,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

;exten => 8368,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

;exten => 8368,5,Hangup

;

;; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:

;exten => 8369,1,Playback(sip-silence)

;exten => 8369,2,AGI(agi://127.0.0.1:4577/call_log)

;exten => 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)

;exten => 8369,4,AGI(VD_amd.agi,${EXTEN})

;exten => 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

;exten => 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)

;exten => 8369,7,Hangup

;

;; VICIDIAL auto-dial reminder script

;exten => 8372,1,Playback(sip-silence)

;exten => 8372,2,AGI(agi://127.0.0.1:4577/call_log)

;exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})

;exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})

;exten => 8372,5,Hangup

;

;; VICIDIAL SURVEY transfer script AMD with Load Balanced:

;exten => 8373,1,Playback(sip-silence)

;exten => 8373,2,AGI(agi://127.0.0.1:4577/call_log)

;exten => 8373,3,AMD(2000|2000|1000|5000|120|50|4|256)

;exten => 8373,4,AGI(VD_amd.agi,${EXTEN})

;exten => 8373,5,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)

;exten => 8373,6,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)

;exten => 8373,7,Hangup

SUBPHASE 6.3: adding entries to your MySQL "asterisk" database for vicidial

applications

We need to add a few initial values to the vicidial tables in the "asterisk"

database in order to start setting up the vicidial dialer system for use.

#### REMOVED, not necessary if you run the first_server_install.sql file above

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SUBPHASE 6.4: setting up asterisk and helper applications for startup

1. Make several entries in the rc.local of your system:

- on the command line type:

- vi /etc/rc.d/rc.local

- add the following entries(here's what we used):

# OPTIONAL enable ip_relay(for same-machine trunking and blind monitoring)

# /usr/share/astguiclient/ip_relay/relay_control start 2>/dev/null 1>&2

# Disable console blanking and powersaving

/usr/bin/setterm -blank

/usr/bin/setterm -powersave off

/usr/bin/setterm -powerdown

### start time server

/usr/local/bin/ntpdate -u ntp.myfloridacity.us

/usr/sbin/ntpd

### start up the MySQL server

/usr/local/mysql/bin/mysqld_safe --user=mysql --skip-name-resolve --skip-host-cache &

### start up the MySQL 4.1.X server (with old passwords)

/usr/local/mysql/bin/safe_mysqld --old-passwords --skip-name-resolve --skip-host-cache &

### start up the apache web server

/usr/local/apache2/bin/apachectl start

### roll the Asterisk logs upon reboot

/usr/share/astguiclient/ADMIN_restart_roll_logs.pl

### clear the server-related records from the database

/usr/share/astguiclient/AST_reset_mysql_vars.pl

### load digium tormenta 4xT1 drivers into system

modprobe zaptel

modprobe wct1xxp

/sbin/ztcfg -vvvvvvvvvvvv

### sybsys local login

touch /var/lock/subsys/local

### sleep for 20 seconds before launching Asterisk

sleep 20

### start up asterisk

/usr/share/astguiclient/start_asterisk_boot.pl

- you are done

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SUBPHASE 6.5: setting up astguiclient scripts for continuous running

1. Make several entries in the crontab of your system:

- on the command line type:

- cd /usr/share/astguiclient

- crontab -e

- add the following entries(here's what we used):

### recording mixing/compressing/ftping scripts

0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl

#0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_VDonly.pl

1,4,7,10,13,16,19,22,25,28,31,34,37,40,43,46,49,52,55,58 * * * * /usr/share/astguiclient/AST_CRON_audio_2_compress.pl --GSM

#2,5,8,11,14,17,20,23,26,29,32,35,38,41,44,47,50,53,56,59 * * * * /usr/share/astguiclient/AST_CRON_audio_3_ftp.pl --GSM

### keepalive script for astguiclient processes

* * * * * /usr/share/astguiclient/ADMIN_keepalive_ALL.pl

### kill Hangup script for Asterisk updaters

* * * * * /usr/share/astguiclient/AST_manager_kill_hung_congested.pl

### updater for voicemail

* * * * * /usr/share/astguiclient/AST_vm_update.pl

### updater for conference validator

* * * * * /usr/share/astguiclient/AST_conf_update.pl

### flush queue DB table every hour for entries older than 1 hour

11 * * * * /usr/share/astguiclient/AST_flush_DBqueue.pl -q

### fix the vicidial_agent_log once every hour

33 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl

### updater for VICIDIAL hopper

* * * * * /usr/share/astguiclient/AST_VDhopper.pl -q

### adjust the GMT offset for the leads in the vicidial_list table

1 1,7 * * * /usr/share/astguiclient/ADMIN_adjust_GMTnow_on_leads.pl --debug

### reset several temporary-info tables in the database

2 1 * * * /usr/share/astguiclient/AST_reset_mysql_vars.pl

### optimize the database tables within the asterisk database

3 1 * * * /usr/share/astguiclient/AST_DB_optimize.pl

## adjust time on the server with ntp

30 * * * * /usr/local/bin/ntpdate -u ntp.myfloridacity.us 2>/dev/null 1>&2

### VICIDIAL agent time log weekly and daily summary report generation

2 0 * * 0 /usr/share/astguiclient/AST_agent_week.pl

22 0 * * * /usr/share/astguiclient/AST_agent_day.pl

### VICIDIAL campaign export scripts (OPTIONAL)

#32 0 * * * /usr/share/astguiclient/AST_VDsales_export.pl

#42 0 * * * /usr/share/astguiclient/AST_sourceID_summary_export.pl

### remove old recordings more than 7 days old

#24 0 * * * /usr/bin/find /var/spool/asterisk/monitorDONE -maxdepth 2 -type f -mtime +7 -print | xargs rm -f

### remove old vicidial logs and asterisk logs more than 2 days old

28 0 * * * /usr/bin/find /var/log/astguiclient -maxdepth 1 -type f -mtime +2 -print | xargs rm -f

29 0 * * * /usr/bin/find /var/log/asterisk -maxdepth 3 -type f -mtime +2 -print | xargs rm -f

30 0 * * * /usr/bin/find / -maxdepth 1 -name "screenlog.0*" -mtime +4 -print | xargs rm -f

- once your system starts up you can attach to the screen running asterisk by

typing "screen -r " find which screen by typing "screen -r" and

looking for the lowest screen number. Then to detach again from the screen

while you are in it type 'Ctrl+a' then 'd'

- you are done

NOTES:

- The AST_agent_day.pl and AST_agent_week.pl scripts create an ASCII fixed-length report of all agent activity on the system

- The AST_VDsales_export.pl script allows for the exporting(into several different formats) of specified vicidial_list data based on status and campaign as well as inbound group

- The AST_sourceID_summary_export.pl script summarizes leads in the system by source_id and generates a text report

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SUBPHASE 6.6: adding test leads to the VICIDIAL database and configuring a

VICIDIAL campaign and users

First we will add a few test leads to the vicidial_list table so that we can

test our system. There is also an application included with the distribution

that will accept a delimited file of leads placed in the /usr/share/astguiclient/VICIDIAL/LEADS_IN/

directory and load it into the database automatically(VICIDIAL_IN_new_leads_file.pl

[a sample lead file in the proper format is included with this release:

- test_VICIDIAL_lead_file.txt])

If you want to use the lead import script I suggest looking at the code to make

sure it is entering what you want it to. We are not going to go over that

particular script in this document.

Also, there is a web-based lead loader that was made available as of the 1.1.1

release and is accessible from the VICIDIAL admin.php web page(click on the

"LOAD NEW LEADS" link at the top of the admin page). To get to this page you

must have permissions in the vicidial_user table(Load Leads set to 1) .

Instructions on it's use are included on the page through the help question

mark link.

NOTE: in PHP you must have "fileuploads" enabled for this page to work.

NOTE: it is important to have your proper country code in the phone_code field of your leads so that the GMT offset encodding will properly code the time zones for your leads. For the USA/Canada/Caribbean this would be 1. For the UK this is 44 and Mexico is 52 and so on.

Second we need to add the disposition statuses into the system, all of these

queries are below:

(Note: you may want to replace 7275551212 with a real number to test in these

records)

#### REMOVED, not necessary if you run the first_server_install.sql file above

Now that the sample leads and disposition codes have been entered, we can go

into the VICIDIAL administration website and set up our campaigns, lists and users.

But first let's make sure that they have the right GMT offset:

run this on the command line:

- /usr/share/astguiclient/ADMIN_adjust_GMTnow_on_leads.pl --postal-code-gmt

3. Enter the astguiclient administration page:

http://10.10.10.15/vicidial/admin.php

(use the username and password created when we entered a record into the

vicidial_users table in SUBPHASE 6.1, In our case this is 6666 and 1234)

NOTE: if you click on the Logout button you must leave the user/pass empty and click OK

- Now that you are logged into the vicidial administration system we can add

new user entries for each of the new users and enter new campaigns and new

lists.

- The first step is to enter your new users, Click on the ADD A NEW USER

and fill in the appropriate information for each now user you want to

add.

- Next, you need to create a new campaign, click on the ADD A NEW

CAMPAIGN link and fill in what you want the campaign to be called as

well as a description

- Next, you need to define a new list, click on the ADD A NEW LIST link

and fill in what you want the list to be called as well as a using the

list ID of the leads that we loaded in the previous step "101" and

select the new campaign from the pull-down menu that we just created.

- Now that you have created your list, make it active by changing active to Y

- now modify your campaign ang change the first status to be called to

NEW and submit. Now your system is ready to dial.

- you are done

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SUBPHASE 6.7: VICIDIAL remote agents:

With v1.0 of VICIDIAL we have the ability to use a simple web form to give

remote agents a way to receive calls to whatever number they happen to be at,

and they can view/edit call details and see a call log all through a web page

(vdremote.php) or http://10.10.10.15/vicidial/vdremote.php on this installation.

Remote Agents is only recommended for inbound calls because of the extra time

needed to dial a number out to transfer the call to. To set up remote agents,

just go to the vicidial admin.php page and ADD NEW REMOTE AGENTS(Make sure the

userID start also has a user login so they can get to the vdremote page). You

will see that you can set up a remote agent entry to take multiple lines if you

wish meaning that, for example, if you need to send all of your calls to another

location because of a massive snowstorm(and none of your agents showed up at work)

you just log in that remote agent record with say 10 lines and then all of those

calls will be directed to the same number you set up for the remote agent. Then

again you could just get your agents to log in from home if they have a phone

and computer

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SUBPHASE 6.8: astGUIclient web-only client:

With 1.1.1 release of astguiclient we have completely rewritten the astGUIclient

client app in AJAX(PHP/Javascript/XMLHTTPRequest) to enable a full, real-time

GUI interface using only a web browser. The browser requirements for this are:

- Firefox 0.9 or greater (Firefox 1.0.7 is the recommended browser)

- Mozilla 1.7 or greater

- Netscape 8 or greater

- Opera 8.5 or greater

- Microsoft Internet Explorer 6.0

This new version also has more flexibility and functionality than the perl/Tk

version as well as being prettier. We have successfully tested this on many

platforms and in remote locations. It functioned wonderfully off-site with one

of our IAX hardphones and offers a lot of promise for road warriors who need

a lot of options on their phone usage like conferencing and a detailed call log.

To log into this app you will need a login setup in the vicidial_users table

with a user_level of 1 or greater as well as an entry for the phone you are

using in the phones table. You will first get a login prompt for the vicidial

for that phone entry. From there the app should display and you will see the

MAIN screen with your phone information, voicemail display and your inbound/

outbound phone call log.

The example web page you would go to on this installation would be:

http://10.10.10.15/agc/astguiclient.php

The inbound log and callerID popup is dependant on having a call_inbound.agi

entry in your dialplan before you phone is dialed(see subphase 6.2 step 2)

Another thing to note is that you can have the agc folder(with the .php files

in it) copied to multiple web servers, you just need to make sure that the

MySQL database connection works (check the settings in the dbconnect.php file

that is in the agc directory). We have had astguiclient.php running on 3

separate web servers for the same DB server and Asterisk server. This is an

easy way to allow for auto failover and/or redundancy. Also, this client will

work over SSL connections(https) for encrypted communications with the server.

New in astGUIclient release 1.1.7 is multi-language support. multi-language

versions of web-clients and admin pages are available in the LANG_www directory

and can be unzipped into your webroot directory.

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SUBPHASE 6.9: VICIDIAL web-only client:

NOTE: There is a VICIDIAL Agent manual available from http://www.eflo.net

With 1.1.6 release of astguiclient we have finished the rewrite of the VICIDIAL

client app in AJAX(PHP/Javascript/XMLHTTPRequest) to enable a full, real-time

GUI interface using only a web browser like we have done with astGUIclient.

The browser requirements for this are:

- Firefox 0.9 or greater (Firefox 1.0.7 is the recommended browser)

- Mozilla 1.7 or greater

- Netscape 8 or greater

- Opera 8.5 or greater

- Microsoft Internet Explorer 6.0

This version is fully functional and has been tested in our production

call center with no problems.

To log into this app you will need a login setup in the vicidial_users table

with a user_level of 1 or greater as well as an entry for the phone you are

using in the phones table. You will first get a login prompt for the vicidial

for that phone entry. From there the app should display and you will see the

VICIDIAL screen with your phone information.

The example web page you would go to on this installation would be:

http://10.10.10.15/agc/vicidial.php

One more feature that the VICIDIAL web-client offers is the ability to set up

an EXTERNAL phone extension in the astguiclient admin section so that you can

have agents log in to vicidial.php wherever they have access to a phone with

an external phone number and a web browser. To do this follow these steps:

- "ADD PHONE" in the admin.php web page and enter whatever name you want

- For the dialplan number field put in the full digits that you would dial from

the Asterisk server to get to that agent's external phone(with 91 if used)

- For the Protocol select EXTERNAL

- make sure the agent knows the login and password set for this phone entry.

Then the agent will go to the vicidial.php page and enter in their phone

few seconds, and they are logged in and ready to take calls.