ATCOM IPPBX IP-2G4A User manual

ATCOM®IPPBX IP-2G4A
Product Guide
Version: 1.0
2010-10-14

www.atcom.cn 1
Content
CONTACT ATCOM ...............................................................................................................................3
CHAPTER 1 THE INTRODUCTION OF IP-2G4A......................................................................................4
CHAPTER 2 ACCESS TO THE IP-2G4A ................................................................................................... 6
2.1 WEBPAGE ACCESS BY BROWSER .............................................................................................................6
2.2 SUPPORT SSH PROTOCAL ...................................................................................................................... 7
2.3 CONSOLE PORT ACCESS TO IP-2G4A ......................................................................................................7
CHAPTER3 GENERAL OPERATION OF IP-2G4A .................................................................................... 8
CHAPTER 4 CONFIGURE IP-2G4A BY WEB GUI .................................................................................. 10
4.1 SYSTEM STATUS ............................................................................................................................10
4.2 CONFIGURE HARDWARE..............................................................................................................10
4.3 TRUNKS ...........................................................................................................................................11
4.3.1 Create Analog Trunks ............................................................................................................12
4.3.2 VoIP Trunks ............................................................................................................................13
4.4 OUTGOING CALLING RULES .................................................................................................................13
4.5 DIAL PLANS ......................................................................................................................................15
4.6 USERS .............................................................................................................................................15
4.7 RING GROUPS................................................................................................................................... 17
4.8 MUSIC ON HOLD................................................................................................................................17
4.9 CALL QUEUES ...................................................................................................................................18
4.10 VOICE MENUS ..............................................................................................................................19
4.11 TIME INTERVALS .............................................................................................................................. 21
4.12 INCOMING CALLING RULES ................................................................................................................21
4.13 VOICEMAIL.....................................................................................................................................22
4.14 CONFERENCING...............................................................................................................................24
4.15 FOLLOW ME...................................................................................................................................26
4.16 DIRECTORY .....................................................................................................................................27
4.17 CALL FEATURES ...............................................................................................................................27
4.18 VOICEMAIL GROUPS ........................................................................................................................28
4.19 VOICE MENU PROMPTS .................................................................................................................... 28
4.20 SYSTEM INFO ..................................................................................................................................28
4.21 BACKUP .........................................................................................................................................29
4.22 ACTIVE CHANNELS ...........................................................................................................................29
4.23 OPTIONS........................................................................................................................................29
CHAPTER 5 AN APPLICATION CASE OF IP-2G4A ................................................................................ 31
5.1 HOW TO MAKE INTERNAL CALLS THROUGH IP-2G4A............................................................................... 32
5.1.1 Access to the Web Page of IP-2G4A by Browser.................................................................... 32
5.1.2 Add up Users from Web Page of IP-2G4A..............................................................................32
5.1.3 Register a SIP user 6001 in AT610..........................................................................................34

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5.2 HOW TO COMMUNICATE WITH OUTSIDE THROUGH PSTN .........................................................................35
5.2.1 Create an Analog Trunk ......................................................................................................... 35
5.2.2 Create an Outgoing Calling Rule ........................................................................................... 35
5.2.3 Selected the Outgoing Calling Rules in a Dial Plan ................................................................36
5.2.4 Create a User.........................................................................................................................36
5.2.5 Create Incoming Calling Rules ...............................................................................................37
5.3 HOW TO CALL EACH OTHER DIRECTLY FROM DIFFERENT NETWORK SEGMENT...........................37
5.4 HOW TO CALL THROUGH VOIP TRUNK...................................................................................................39
5.4.1 Call from IP-2G4A to IP04......................................................................................................39
5.4.2 Call from IP04 to IP-2G4A......................................................................................................41
5.5 VOICEMAIL.......................................................................................................................................42
5.6 HOW TO REALIZE THE IVR ...................................................................................................................42
5.6.1 Upload Voice Menu Prompts .................................................................................................43
5.6.2 Create Voice Menu ................................................................................................................43
5.6.3 Add Incoming Calling Rules ...................................................................................................45
5.7 CONFERENCE ....................................................................................................................................46
5.8 RING GROUPS................................................................................................................................... 46
5.9 AGENTS ...........................................................................................................................................47
5.9.1 Create Users as Agents..........................................................................................................47
5.9.2 Create a Call Queue...............................................................................................................48
5.10 FOLLOW ME...................................................................................................................................49
5.11 GROUP CALL PICKUP ........................................................................................................................50
ACRONYMS ...................................................................................................................................... 52
GLOSSARY ........................................................................................................................................ 54
REFERENCE....................................................................................................................................... 55

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Contact ATCOM
The Introduction of ATCOM
Founded in 1998, ATCOM technology has been always endeavoring in the R&D and
manufacturing of the internet communication terminals. The product line ofATCOM includes IP
Phone, USB Phone, IP PBX, VoIP gateway andAsterisk card.
Contact Sales:
Address
District C, east of 2nd floor, #3, Crown industry buildings, Chegongmiao
Industry area, Futian district, Shenzhen, China
Tel
+(86)755-23487618
Fax
+(86)755-23485319
E-mail
[e-mail]
Contact Technical Support:
Tel
+(86)755-23481119
E-mail
[e-mail]
Website Address: http://www.atcom.cn/
ATCOM Wiki Website: http://www.openippbx.org/index.php?title=Main_Page
Download Center: http://www.atcom.cn/download.html

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Chapter 1 the Introduction of IP-2G4A
1. Overview of the IP-2G4A
The IP-2G4A is a complete Asterisk Appliance with combination of GSM and Ananlog channels.
It is an embedded open source Linux system with built-in SIP/IAX2 proxy server and NAT
functions. It provides a solid, uniform platform for Mobile and VoIP communications.
Targeting for SOHO user and SMB market with an easy to use graphical interface, ATCOM GSM
IP PBX provides a cost-saving solution on their telecommunication/data needs. With these devices,
company with branch offices in different countries can be easily combined together to work like a
virtual single office through internet, GSM and PSTN network.
2．Hardware
CPU: 400MHz Blackfin 532 Chip
2 x GSM ports and four analog ports
NAND flash 256 M
SDRAM 64M
3．System
Open Source uClinux
4. Function features
PSTN, GSM,ISDN
Support g711/g729/gsm codec
Voicemail
Voicemail groups
3-way Calling
Conferencing
Follow Me
Call Feature
In directory
Call Waiting
Call Queues
Pickup Group
Ring Group
Is Agent
Music On Hold
Voice Menus
Voice menus Prompts
Time intervals
Backup
Update

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5．Applications
SOHO/SMB telephony system
Hosted service
IVR system
6. Interface
1 X RJ45 port.
1 X Power port.
1 X RS232 port.
4 X FXO/FXS ports.
2 X GSM channels.
7．Measurement and Weight
Inner box
225 * 120 * 30mm
G.W./unit
0.79KG
Carton MEAS
456 * 442 * 362 mm
Units per Carton
21 units/ CTN
G.W./CTN
18 KG/CTN
8．Package
Item
Quantity
IP-2G4A
1
RS232 module
1
Power Adapter
1
Manual (disk)
1
For the usage of IP-2G4A in VoIP field, you can refer to the following network topology.

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Chapter 2 Access to the IP-2G4A
You need a PC to access to the IP-2G4A, there are four ways for you to access the IP-2G4A:
1. Web page access by browser
2. SSH access by putty
3. Access by browser with Fallback IPAddress
4. Console port access by RS232 console cable
In order to access to IP-2G4A by the first three ways, you have to check that if your network
connection between IP-2G4A and PC is OK. If you do not have network connection between
IP-2G4A and PC, you can try to use the last way to access to IP-2G4A and change the IP address
for IP-2G4A.
2.1 WebPage Access by Browser
It is the most convenient and common way to access the IP-2G4A, you just need to open your
browser and input the IP address of IP-2G4A WAN port (the default IP address is 192.168.1.100).
You’d better use Firefox instead of IE, because there are compatible issues.
Then input the default Username: admin; Password: atcom .

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2.2 Support SSH protocal
Logging into IP-2G4A by SSH, you can configure IP-2G4A by Linux command.
2.3 Console Port Access to IP-2G4A
If you do not have network connection between IP-2G4A and PC, you can try to access to
IP-2G4A by console port. Please try to do as the following steps:
1. Connect the console port of IP-2G4A to your PC’s console port with RS232 console cable.
2. Run your HyperTerminal, and set up the console port like the following:
Bits per second: 115200
Data bits : 8
Parity: None
Stop bits: 1
Flow control: None
3. Change the IPAddress by HyperTerminal
The default IP address of IP-2G4A is 192.168.1.100. Your network may have a different IP
address segment such as 192.168.10.xx. In this situation, you cann’t access to IP-2G4A by putty
and browser if you do not change the IP-2G4A IP address. So you have to change the IP address
for IP-2G4A by HyperTerminal to make it in the same network segment as your LAN.
After you have accessed to IP-2G4A by HyperTerminal, please use the following command to
change the IP address for IP-2G4A.
root:~> ifconfig eth0 192.168.1.151(the IP address what you want to set for IP-2G4A)
By this way, the IP address you set for IP-2G4A is temporary, it will recover to the original default
IP address after rebooting. If you want to give a static and permanent IP address for IP-2G4A, you
can try to set it in web GUI, for detail steps please refer to chapter 3.

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Chapter3 General Operation of IP-2G4A
1. Backup
When you log in the web of IP-2G4A, Click on Backup, you can see the button of Create
New Backup,then you can Backup the current system.
2. System Update
When you log in the web of IP-2G4A, click on Options Advanced Options Show
Advansed Options , After click on Show Advanced Options in the web, you can see the
advanced options in the vertical menu on the left of the main page. Click on Firmware
update , you can see the following parameters in the table.
Parameter Name
Description
Type
Default
HTTP URL
The http path of the firmware file
Textbox
Null
TFTP Server
The IP address of TFTP Server where the
firmware file in.
Textbox
Null
File Name
Specify the name of your uImage-md5
firmware file, make sure to use md5 version
only
Textbox
Null
Reset Configs
Select this box if you wish to reset to factory
defaults. This will ensure a clean update, and is
highly recommended
Selected
Not selected
If you want to upload sound file, upload backup files and so on , you can refer to the link:
http://www.atcom.cn/downloads/IPPBX/ATCOM%20IPPBX%20Series%20Product%20Upgr
ade%20Guide-V1.0-EN.pdf
3. Network
After click on OptionsAdvanced OptionsShow Advanced Options, please select
Network Settings option from the vertical menu on the left of main page. You can set IP
address, Subnet mask, Gateway, DNS what you want like the following:

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Please click save button in your page to save your setting and reboot the IP-2G4A.
Attention: you need configure the IP address, Subnet mask, Gateway and DNS at WAN
Interface so that the network connects successfully. The option of LAN Interface is used for
Routing functions, here you needn’t configure it.

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Chapter 4 Configure IP-2G4A by Web GUI
4.1 System Status
In the system status screen, it displays the functions you configured, such as: trunks, extensions,
conference and so on. The following table is the options description of trunks.
Name
Description
Status
The register status of trunks
Trunk
The name of trunks
Type
The type of trunks
Username
The username of SIP/IAX trunk
Port/Hostname/IP
IP Address/port
1.The register status of trunks include three kinds: Unregistered, Request Sent, Registered.
2.The type of trunks : VoIP trunk including SIP and IAX;Analog trunk; Service Provider.
The parameter of extensions in the following table:
Name
Description
Extension
The status of users
Name/label
The name of users
Status
Display voice message
Type
SIP users/IAX users/Analog users
1. There are four kinds status of users, when the light of “Extension” list displays gray , means the
user does not register that is Unavailable; when the light of “Extension” list displays green ,
means the user is Free; when the light of “Extension” list displays orange , means the user is
Ringing; when the light of “Extension” list displays red , means the user is Busy.
2. Status: This parameter displays if other users leave messages, Messages : 0/0, the figure front of
“/ ”displays the new messages amount; the figure behind of displays the old messages
amount.
4.2 Configure Hardware
In the configure hardware page, it includes the following components: analog hardware, tone
region, advanced settings. Pay attention that some browsers do not display the configure, it is
unimportant.
Analog Hardware
When you boot the IP-2G4A, which will detect the FXO and FXS modules automatically, the
analog hardware component displays the modules which are detected correctly.

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Name
Description
Type
Default
Tone Region
Select the tone region according to your
country, if it does not have your country’s
name in the dropdown list, please ask your
service operator which kind of tone region is
used in your area
ComboBox
United
Status/North
America
Module Name
The name of Module
Textbox
wctdm24xxp
Opermode
Specifies On Hook Speed, Ringer Impedance,
Ringer Threshold, current
Limiting ,TIP/RING voltage adjustment,
minimum Operational Look Current and so
on. Please choose your country or your nearest
neighboring country
ComboBox
USA
a-law override
Specifies the codec to be used for analog line.
ComboBox
ulaw
fxs honor mode
This option allows the user to determine if
they would like opermode characteristics
applied to trunk(FXO) modules only, or both
trunk (FXO) and station(FXS) modules.
ComboBox
FXO modules
boostringer
This option allows the user to define whether
they require normal ringing voltage(40v) or
maximum ringing voltage(89v) or analog
phones attached to station(FXS) modoules
ComboBox
nomal
fastringer
This option sometimes used in conjunction
with the Low Power Option ,allows the user to
increase the ringing speed to 25HZ
ComboBox
nomal
lowpower
This option generally used in conjunction with
the Fast Ringer Option ,allows the user to set
the peak voltage during Fast Ringer Operation
to 50V.
ComboBox
nomal
ring detect
This option allows the user to choose from
normal ring detection or a full wave detection
ComboBox
standard
MWI mode
This option allows the user to specify the type
of Message Waiting indicator detection to be
done on trunk(FXO) interfaces
ComboBox
none
4.3 Trunks
To receive calls from PSTN and make calls to the outside world, you have to use trunk. Please
select the Trunks option from the vertical menu on the left of the main page.

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4.3.1 Create Analog Trunks
Analog trunk is associated with FXO port, and it will call outside by PSTN line. Click on New
Analog Trunk , then we can see the parameters which are in the following table in the web.
Name
Description
Type
Default
Channels
Display the FXO or GSM modules
selected
no select
Trunk Name
The name you want to set for the trunk
Textbox
null
Busy Detection
Busy detection is used to detect far end hang
up or for detecting busy signal.
Boolean
Yes
busycount
If Busy Detection is enabled,it is also possible
to specify how many busy tones to wait for
before hanging up.
Int
3
Ring Timeout
Thrunk(FXO) devices must have a timeout
to determine if there was a hangup before the
line was answered.
Int
8000
answeronpolarit
yswitch
If this option is enabled, the reception of a
polarity reversal will mark when a outgoing
call is answered by the remote party.
Boolean
no
hanguponpolarit
yswitch
In some countries ,a polarity reversal is used
to single the disconnect of a phone line.
Boolean
no
Use CallerID
Enabling this option enabled CallerId
detection.
Boolean
yes
Caller ID Start
This option allows one to define the start of a
CallerID Signal.
ComboBox
Ring
CallerID
This option allows the lines to report the
Caller ID string as received from the telco, or
as a fixed value by using the custom option.
select box
As Received
Pulse Dial
If this option is enabled ,pulse mode dialing
instead of DTMF,wil be enable.
Boolean
No
CID Signalling
This option defines the type of caller ID
signaling to use :bell,v23,v23_jp,or dtmf.
ComboBox
Bell-USA
Flash Timing
Flash Time defines the time ,in
millseconds,that is generated for a flash
operation.
Textbox
750
Receive Flash
Timing
Flash Time defines the time,in milliseconds,
that is generated for a flash operation.
Textbox
1250

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1.Trunk name: unique label to help you identify the trunk when listed in outgoing calling rules
and incoming calling rules.
4.3.2 VoIP Trunks
A VoIP service provider (VSP) that you have signed up with is also a trunk. Via the VoIP trunk you
can dial via the VoIP service to reduce your cost when making international calls. You can set up
the VoIP trunk to make calls to the PSTN or other VoIP network depends on the service you use.
You can also use the VoIP trunk to link headquarter and branch offices for free internal calls.
Click on New SIP/IAX Trunk, the following table is the parameter of VoIP trunk:
Name
Description
Type
Default
Type
You can select SIP or IAX type to meet your
need.
ComboBox
SIP
Provider Name
A unique label to help you identify this trunk
when listed in outbound rules, incoming rules etc.
Textbox
Null
Hostname
The IP Address of the server which you want to
connect
Textbox
Null
Username
the username that your service provider
configured
Textbox
Null
Fromdomain
The domain of the server which you want to
connect
Textbox
Null
Password
the password that your service provider
configured for the user.
Textbox
Null
Contact Ext.
Textbox
s
Insecure Type
The insecure type of the trunk transferring data.
ComboBox
very
1.Notice Provider Name must be unique label , especially do not the same with Username .
2.Insecure Type: insecure=very ; To allow registered hosts to call without re-authenticating
insecure=port ;Allow matching of peer by IP address without matching port number.
insecure=invite; removes the requirement for authentication of incoming INVITE messages.
4.4 Outgoing Calling Rules
Outgoing calling rules is used to route an outgoing call, when you make an external call, which
trunk and what dial-pattern the call used are configured in outgoing calling rules. Please select the
Outgoing Calling Rules option, then Click on New Calling Rule button, the parameters of the
Outgoing Calling Rules are in the following table:

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Name
Description
Type
Default
Calling Rule Name
The name of the Calling rule
Textbox
Null
Pattern
The dialing rule
Textbox
Null
Send to Local
Destination
If this option is checked and Destination is
defined, calls matching the specified pattern
may be sent to a local extension."
selected
no select
Destination
Choose the Local
Destionation:User/VoiceMenu/Hungup…
ComboBox
Null
Use trunk
Defines the Trunk that calls, matching the
specified pattern, will be placed through.
ComboBox
Null
Strip
Allows the user to specify the number of digits
that will be stripped from the front of the
dialing string before the call is placed via the
trunk selected in \"Use Trunk.\" One might.
Textbox
Null
Prepend these
digits
Allows the user to specify digits that are
prepended before the call is placed via the
trunk. If a user's trunk required 10 digit
dialing, but users were more comfortable
performing 7 digit dialing, this field could be
used to prepend a 3 digit area code to all 7 digit
strings before they are placed to the trunk. User
may also prepend a 'w' character for analog
trunks to provide a slight delay before dialing
Textbox
Null
Use Failover
Trunk
Failover trunks can be used to make sure that a
call goes through an alternate route, when the
primary trunk is busy or down If "Use Failover
Trunk" is checked and "Failover trunk" is
defined, then calls that cannot be placed via the
regular trunk may have a secondary trunk
defined. If a user's primary trunk is a VoIP
trunk, but one wants calls to use the PSTN
when the VoIP trunk isn't available, this option
is a good idea.
selected
no select
Fail over trunk
Choose the trunk
ComboBox
ComboBox
1. Pattern: X ... Any Digit from 0-9; Z ... Any Digit from 1-9; N ... Any Digit from 2-9;
[12345-9] ... Any Digit in the brackets (in this example, 1,2,3,4,5,6,7,8,9); .... Wildcard, matches
anything remaining; i.e. _9011. Matches anything starting with 9011 (excluding 9011 itself); ! ...
Wildcard, causes the matching process to complete as soon as it can unambiguously determine
that no other matches are possible. For example, the extension _NXXXXXX would match
normal 7 digit dialings, while _1NXXNXXXXX would represent a three digit area code plus
phone number, proceeded by a one.
2. Strip: Allows the user to specify the number of digits that will be stripped from the front of the
dialing string before the call is placed via the trunk selected in Use Trunk. For example, want
users to dial 9 before their long distance calls; however one does not dial 9 before those calls

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are placed onto analog lines and the PSTN, so one should strip 1 digit from the front before the
call is placed.
3. The way of outgoing calling rules works:
Every time you dial a number, asterisk will do the following in strict order:
• Examine the number you dialed.
• Compare the number with the pattern that you have defined in your first outgoing rule and if
matches, it will initiate the call using that trunk. If it does not match, it will compare the number
with the pattern that you have defined in the second outgoing rule and so on.
• Pass the number to the appropriate trunk to make the call.
4.5 Dial Plans
A DialPlan is a set of Calling Rules that can be assigned to one or more users. Please select the
Dial Plans option, Click on New DialPlan button, the following table displays the parameters of
Dial Plans .
Name
Description
Type
Default
DialPlan Name
The name of DialPlan, which is a unique label to
help you identify the dial plan
Textbox
DialPlan1
Include Outgoing
Calling Rules
Select outgoing call rule which you use
selected
Not seclect
Include Local
Contexts
Local context is used for general using
configuration.
check box
Select all
4.6 Users
Users component is used to add or remove Analog, SIP, IAX extension.
Click on Create New User button in the web of IP-2G4A, you can create SIP/IAX User and
Analog User.
Name
Description
Type
Default
Extension
The numbered extension
Textbox
6001
Name
A character-based name for this user
Textbox
Null
DialPlan
DialPlans are sets of calling rules and can be managed
form the \"Dial Plans\" panel
ComboBo
x
Null
CallerID
The Caller ID (CID) string used when this user calls
another internal user.
Textbox
Null
OutBound
CallerID
Caller ID that would be applied for out bound calls
from this user. Note that your ability to manipulate
your outbound Caller ID may be limited by your VoIP
provider.
Textbox
Null
Enable
Check this box if the user should have a voicemail
Selected
Not

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Voicemail for
this User
account
selected
VoiceMail
Access PIN
code
Voicemail Password for this user
Textbox
Null
Mailbox
Voicemail Mailbox for this user
Textbox
Null
Email Address
The e-mail address for this user
Textbox
Null
SIP
Check this option if the User or Phone is using SIP or
is a SIP device
selected
selected
IAX
Check this option if the User or Phone is using IAX or
is an IAX device
selected
selected
Analog
Station
If this user is attached to an analog port on the system,
please choose the port number here
ComboBo
x
Null
Codec
Preference
Choose priority codec
ComboBo
x
u-law/GS
M
NAT
Try this setting when Asterisk is on a public IP,
communicating with devices hidden behind a NAT
device (broadband router). If you have one-way audio
problems, you usually have problems with your NAT
configuration or your firewall's support of SIP+RTP
ports.
selected
selected
Can Reinvite
By default, Asterisk will route the media steams from
SIP endpoints through itself. Enabling this option
causes asterisk to attempt to negotiate the endpoints to
route the media stream directly, bypassing asterisk. It
is not always possible for asterisk to negotiate
endpoint-to-endpoint media routing.
selected
Not
selected
DTMF Mode
Set default dtmfmode for sending DTMF. info : SIP
INFO messages;inband : Inband audio (requires 64 kbit
codec -alaw, ulaw); auto : Use rfc2833 if offered,
inband otherwise.
ComboBo
x
rfc2833
3-Way Calling
Check this option if the User or Phone should have
3-Way Calling capability.
selected
Not select
In Directory
Check this option if the user is to be listed in the
system telephone directory.
selected
Not select
Call Waiting
Check this option if the User or Phone should have
Call-Waiting capability
selected
Not select
Is Agent
Check this option if this User or Phone is a Call Queue
Member (Agent)
selected
Not
selected
Pickup Group
If a user called A and another user called B in the same
group,A can pick up the phone taking the place of B.
selected
Not
selected
1. Analog Station: When you want to createAnalog Users, please choose the FXS ports.
2. Codec Preference: Support g711u-law/g711a-law/g729/GSM

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3. Attension: in the textbox of Extension, the value you set is limited to a range, you can adjust
the range in the Options option to meet your requirement.
4.7 Ring Groups
Define Ring groups to dial more than one extension simultaneously, or to ring more than one
phone sequentially. This feature may also be called Hunt groups.
Please select the Ring Groups option from the vertical menu on the left of the main page, then you
can get the following screen: insecure=invite
Name
Description
Type
default
Ring Group
Name
Ring group name use in pbx
Str*
Extension for
this ring group
Ring group No.，dial the No. if you want to join , change
boundary value in options
Int
6400
Ring Group
Members
The ring group of numbers
{EXT1,EX
T2,EXT3,
…}
Available
Users
The entire Users
{EXT1,EX
T2,EXT3,
…}
Strategy
Ring all simultaneously:
Ring in order
{ Ring in
Order ,ting
all
Extensions
}
Ring in
Order
Seconds to
ring each
member
Seconds to ring each member
Time
20
If not
answered Goto
If not answered go to, hang up: hang up the calling channel。
Operator：Go to operator 。Extension: a call to user.
Voicemail: Go to IVR 。Conference: join a conference.
Call queue: Go to a call queue.
{Hang-up,
Operator…
.}
Hang up
1. ring group application: Dial(channel type/${EXTEN}| channel type/${EXTEN}|20|i)
2. ring group up after please a call
3. non-ring if ring group user off hook or non-user registered
4. only one man can connected in coming call
4.8 Music on hold
'Music On Hold' lets you customize audio tracks for different queues, parked calls etc.
Name
Description
Type
default
Upload an 8 KHz
Mono Music file
Support codec: g711a/g711u
Upload
New music on hold
Add a new music on hold
1. Music on hold Dir: /persistent/sounds/moh/

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2. Sounds : LICENSE-asterisk-moh-freeplay-ulaw
LICENSE-asterisk-moh-freeplay-ulaw
fpm-world-mix.ulaw
fpm-world-mix.alaw
fpm-sunshine.ulaw
fpm-sunshine.alaw
fpm-calm-river.ulaw
fpm-calm-river.ulaw
3. Music on hold after holding status Status: busy
4. Music on hold non-rtp stream
4.9 Call Queues
Please select the Call Queues option from the vertical menu on the left of the main page, then you
can get the following screen:
Name
Description
Type
default
Extension
Extension for call queue: may be dialed to reach the call
queue
Int
6500
Name
Name for call queue
Str*
Strategy
Strategy: this option sets the ringing strategy for this
queue, the options are
1. Ring all: ring all available agents simultaneously until
one answers.
2. RoundRbin: Take turns ringing each available agent.
3. LeastRecent: Ring the agent which was least recently
called
4. FewestCalls: Ring the agent with the fewest
completed calls
5. Random: Ring a Random agent
6. RRmemory: RoundRobin with Memoryn Remember
where it left in the last ring pass
{ringall,Ro
undrobin,le
astrecent,F
ewest
calls,Rand
om,Rrmom
ery}
ring all
Music On
Hold
Select the 'Music on Hold' Class for this Queue. 'Music on
Hold' classes can be managed from the the 'Music On
Hold' panel on the left
Choice
default
LeaveWhen
Empty
This option controls whether callers already on hold are
forced out of a queue that has no agents. There are three
options.
Yes: Callers are forced out of a queue when no agents are
logged in.
No: Callers will remain in a queue with no agents.
Strict: Callers are forced out of a queue with no agents
logged in, or if all logged in agents are unavailable.
The default option is Strict.
After a caller has left the queue, a caller will hear a busy
tone and advance to the next calling rule after attempting
to enter the queue
{yes,strict,
No,}
strict
JoinEmpty
This option controls whether callers can join a call queue
that has no agents. There are three options,
Yes: Callers can join a call queue with no agents or only
unavailable agents
No: Callers cannot join a queue with no agents
Strict: Callers cannot join a queue with no agents or if all
agents are unavailable.
{yes,strict,
No,}
no

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TimeOut
How many seconds an Agent's phone will ring before the
Queue tries to ring the next Agent
Time
15
Wrapup
Time
How many seconds after the completion of a call an Agent
will have before the Queue can ring them with a new call.
The default is 0, which is no delay
Time
0
Max Len
How many calls can be queued at once. This count does
not include calls that have been connected with Agents, it
only includes calls that have not yet been connected.
Default is 0, which is no limit. When the limit has been
reached, a caller will hear a busy tone and advance to the
next calling rule after attempting to enter the queue
Int
0
Auto full
Defining this option causes the Queue, when multiple calls
are in it at the same time, to push them to Agents
simultaneously. Thus, instead of completing one call to
an Agent at a time, the Queue will complete as many calls
simultaneously to the available Agents
checkbox
Auto pause
Enabling this option pauses an agent if they fail to answer
a call. This means that the agent is still logged into the
queue, but they will not receive calls from the queue.
Once paused, an agent can unpause by logging into the
queue using the regular agent login extension
checkbox
Report Hold
Time
Enabling this option causes Asterisk to report, to the
Agent, the hold time of the caller before the caller is
connected to the Agent.
checkbox
KeyPress
Events
If a caller presses a key while waiting in the queue, this
setting selects which voice menu should process the key
press
choice
Agent
This selection shows all Users defined as Agents in their
User conf. Checking a User here makes them a member
of the current Queue
checkbox
1. Call queue application: Queue(${EXTEN})
2. Change agents status:Login / Login out agents in System Info
3. Hear the music if all agents are busy, until non-conversation busy.
4.10 Voice Menus
Like most organization, we would like to redirect all of the incoming calls automatically. The
voice menu is very handy for these sorts of things. The system should allow callers to make the
selection according to the voice menu.
Name
Description
Type
default
Name
A name for the voice menus
Str*
Extension
If you want this Voicemenu to be accessible by dialing an
extension, then enter that extension number
No.
7001
Actions
A sequence of actions performed when a call enters the
menu
Dial plan
script
Add new Step
Add additional steps performed during the menu
Dial plan
script
Allow
KeyPress
Events
Allow key press events will cause the system to listen for
DTMF input from the caller and define the actions that
occur when a user presses the corresponding digit
checkbox
Advance edit
Advance edit for the voice menu
Dial plan
script