Allo.com Nano 2 User manual

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Tables of Contents
1. INTRODUCTION 4
1.1 TYPICAL SETUP OF NANOPBX 5
1.2 NANOPBX TECHNICAL SPECIFICATIONS 5
1.3 HARDWARE SPECIFICATIONS 6
2. GETTING STARTED WITH THE NANOPBX 7
2.1 INSTALLATION 7
2.2 ACCESSING THE GUI (GRAPHICAL USER INTERFACE)8
3. SETTING UP FEATURES 9
3.1 SYSTEM STATUS ERROR!BOOKMARK NOT DEFINED.
3.2 CONFIGURE HARDWARE ERROR!BOOKMARK NOT DEFINED.
3.3 TRUNKS ERROR!BOOKMARK NOT DEFINED.
3.3.1 ANALOG TRUNKS ERROR!BOOKMARK NOT DEFINED.
3.3.2 VOIP TRUNKS ERROR!BOOKMARK NOT DEFINED.
3.4 OUTGOING CALLING RULE ERROR!BOOKMARK NOT DEFINED.
3.5 DIALPLANS ERROR!BOOKMARK NOT DEFINED.
3.6 USERS (USER EXTENSIONS) 36
3.6.1 CREATE NEW USER 27
3.6.2 MODIFY/DELETE SELECTED USER 27
3.7 RING GROUPS ERROR!BOOKMARK NOT DEFINED.
3.8 MOH ERROR!BOOKMARK NOT DEFINED.
3.9 CALL QUEUES ERROR!BOOKMARK NOT DEFINED.
3.9.1 CONFIGURING A QUEUE ERROR!BOOKMARK NOT DEFINED.
3.9.2 AGENT LOGIN SETTINGS ERROR!BOOKMARK NOT DEFINED.
3.10 VOICE MENUS ERROR!BOOKMARK NOT DEFINED.
3.11 TIME INTERVALS ERROR!BOOKMARK NOT DEFINED.
3.12 INCOMING CALLING RULES ERROR!BOOKMARK NOT DEFINED.
3.13 VOICEMAIL ERROR!BOOKMARK NOT DEFINED.
3.13.1GENERAL SETTINGS (ACCESSING RETRIEVING & MANAGING VOICEMAIL) ERROR!BOOKMARK NOT DEFINED.
3.13.2EMAIL SETTINGS FOR VOICEMAIL ERROR!BOOKMARK NOT DEFINED.
3.14 PAGING / INTERCOM ERROR!BOOKMARK NOT DEFINED.
3.14.1GROUP PAGING AND INTERCOM ERROR!BOOKMARK NOT DEFINED.
3.14.2PAGE AN EXTENSION ERROR!BOOKMARK NOT DEFINED.
3.14.3SETTINGS ERROR!BOOKMARK NOT DEFINED.
3.15 CONFERENCING ERROR!BOOKMARK NOT DEFINED.
3.16 FOLLOW ME ERROR!BOOKMARK NOT DEFINED.
3.17 DIRECTORY ERROR!BOOKMARK NOT DEFINED.
3.18 CALL FEATURES ERROR!BOOKMARK NOT DEFINED.
3.19 VOICEMAIL GROUPS ERROR!BOOKMARK NOT DEFINED.
3.20 VOICE MENU PROMPTS ERROR!BOOKMARK NOT DEFINED.

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3.20.1RECORDING VOICEFILES ERROR!BOOKMARK NOT DEFINED.
3.20.2UPLOADING VOICEFILES ERROR!BOOKMARK NOT DEFINED.
3.21 SYSTEM INFO ERROR!BOOKMARK NOT DEFINED.
3.22 BACKUP ERROR!BOOKMARK NOT DEFINED.
3.23 OPTIONS ERROR!BOOKMARK NOT DEFINED.
3.23.1GENERAL PREFERENCES ERROR!BOOKMARK NOT DEFINED.
3.23.2LANGUAGE ERROR!BOOKMARK NOT DEFINED.
3.23.3GENERAL SETTINGS ERROR!BOOKMARK NOT DEFINED.
3.23.4REBOOT ERROR!BOOKMARK NOT DEFINED.
3.23.5ADVANCED OPTIONS ERROR!BOOKMARK NOT DEFINED.
3.24 ASTERISK LOGS ERROR!BOOKMARK NOT DEFINED.
3.25 SIP LOGS ERROR!BOOKMARK NOT DEFINED.
3.26 BILLING INFO ERROR!BOOKMARK NOT DEFINED.
3.27 NETWORK SETTINGS ERROR!BOOKMARK NOT DEFINED.
3.28 FIRMWAREUPGRADE ERROR!BOOKMARK NOT DEFINED.
3.29 OPTIONS >CALL DETAIL RECORDS ERROR!BOOKMARK NOT DEFINED.
3.30 OPTIONS >ACTIVE CHANNELS ERROR!BOOKMARK NOT DEFINED.
3.31 OPTIONS >BULK ADD ERROR!BOOKMARK NOT DEFINED.
3.32 OPTIONS >FILE EDITOR ERROR!BOOKMARK NOT DEFINED.
3.33 OPTIONS >ASTERISK CLI ERROR!BOOKMARK NOT DEFINED.
3.34 OPTIONS >SIP SETTINGS ERROR!BOOKMARK NOT DEFINED.
3.35 OPTIONS >DIAGNOSTIC ERROR!BOOKMARK NOT DEFINED.
3.36 OPTIONS >NETWORK SECURITY ERROR!BOOKMARK NOT DEFINED.
3.37 OPTIONS >PORT CONFIGURATION ERROR!BOOKMARK NOT DEFINED.
4. APPLY CHANGES 55
5. APPENDIX A: MANAGING & HANDLING NANOPBX FEATURES 55
5.1 ATTENDED TRANSFER 55
5.2 BLIND TRANSFER 55
5.3 3-WAY CONFERENCING 56
5.4 CALL FORWARDING 56
5.5 CALL PARKING 56
5.6 CALL HOLD AND RETRIEVE 57
5.7 BLF (BUSY LAMP FIELD)SUPPORT 57
5.8 HARD RESET 57
5.9 MEMORY CARD HANDLING 57
5.10 NPA DIALING 57
6. APPENDIX B: PHONES CONFIGURATION REQUIRED FOR NANOPBX INTEROPERABILITY 58
6.1 X-LITE CONFIGURATIONS 58
6.2 SNOM IP PHONE CONFIGURATION 58
6.3 LINKSYS PAP2 ATA 58

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7. APPENDIX C: GLOSSARY OF TERMS 58
1. Introduction
The NanoPBX is a compact system that puts the rich features of a high-end PBX into the reach of small
businesses. Its built-in voicemail, multi-level auto attendants, remote extensions and sophisticated call
handling features help businesses reduce communications costs, while allowing employees to stay
connected worldwide. Setting up and configuring the NanoPBX is a breeze with the user-friendly GUI and
this document will show you just how easy it is!

Typical setup of NanoPBX
Interconnection Diagram of NanoPBX
NanoPBX Technical Specifications
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PBX Features: Codec: G.711 A-law & μ-law, G.723, G.729
Call Forwarding
Holding and retrieval Fax: T.38, Pass-through
Call Transfer - Attended and Blind
3-way Conference DTMF: RFC 2833, In-band, SIP Info

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Call Parking, Pickup and Retrieval
Remote Voice mail Access System Capacity & Expandability:
Recordable IVR Prompts Supports up to 50 IP Extensions
Multilevel auto attendant with IVR 8 TDM concurrent calls (6 G729)
Intercom 20 IP concurrent calls
Configurable Call Routing
- Least Cost Routing
- Call Route to Multiple Extensions or Targeted User
- Call Hunting - Sequential, Round Robin, Random
TDM + IP scenarios:
8 TDM + 3 IP
4 TDM + 8 IP
0 TDM + 20 IP
Multiple Music on Hold Stacking: Double the capacity of NanoPBX
Call Queue
Direct Inward Dial (DID) routing Internet:
Caller ID Spoofing DHCP server on the LAN side
Busy Lamp Field (BLF) DHCP client on the WAN side
Regional Caller ID & Country Settings STUN/NAT traversal (RFC 3489)
CDR(Caller ID, Time/Date Incoming, Outgoing,
Missed)
SIPv2 - Session Initiation Protocol Version
2 (RFC 3261, 3262, 3263, 3264)
Voice Features: Host Name Config
Echo cancellation (G.168) System Management:
VAD with silence suppression WEB interface accessible from LAN & WAN
CNG Password control
Packet loss concealment Backup and Restore of configuration
Adaptive jitter buffer Software Upgrade
Signal discriminator Factory Reset
Attenuation/Gain adjustment Date & Time Configuration
Hardware Specifications
Telephone Interfaces (FXS/FXO) •Phone 1 – FXS
•Phone 2 – FXS
•Phone 3 – FXS
•Phone 4 – FXS
•Phone 5 – Programmable (FXS / FXO)
•Phone 6 – Programmable (FXS / FXO)
•Line 1 – FXO
•Line 2 – FXO
Network Interfaces One 10/100Mbps WAN Port
One 10/100Mbps LAN Port

System Indicators(LED) •Power (Green)
•1 – Phone1 (Orange & Blue)
•2 – Phone2 (Orange & Blue)
•3 – Phone3 (Orange & Blue)
•4 – Phone4 (Orange & Blue)
•5 – Phone 5 (Orange & Blue)
•6 – Phone 6 (Orange & Blue)
•7 & 8 – Reserved for Internal use
•1 & 2 – Reserved for Internal use
Internal/External Memory: RAM: 64 MB
Flash: 8 Mb
1 GB SD Card Memory
Power Adapter Input: 100V – 240V AC / 47 ~ 63 Hz
Output: +12V DC / 2.5 A
Dimension 175mm (W) x 253mm (L) x 34 mm(H)
(6.88” x 9.96” x 1.33”)
Unit Weight 0.55 kg
Operating Temperature
10°C to 40°C (50°F to 104°F)
Operating humidity
10% to 90%, Non-condensing
Storage temperature 0°C to 50°C (32°F to 122°F)
Storage humidity 5% to 95%, Non-condensing
2. Getting Started With the NanoPBX
Installation
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Step 1: Plug one end of the RJ45 Ethernet
cable into your Router
Step 2: Plug the other end of the RJ45
Ethernet cable into the WAN port of the
NanoPBX
Step 3: Plug the Power Adapter included

into an available power outlet
Step 4: Plug the other end of the Power
Adapter into the “DC-IN” port of the
NanoPBX
Step 5: The NanoPBX will power up, and automatically connect itself to your network via DHCP (which
you can later configure in the SETTINGS > Network Settings section)
Important Note:
•Wait until the POWER and all the six PHONE LED's turn orange and remain stable on the Front Panel
of your NanoPBX.
•Use Straight – through Ethernet cable to connect between the NanoPBX to Router/Switch/PC
Step 6: Configure your NanoPBX according to the instructions below
Accessing the GUI (Graphical User Interface)
Connect an Analog phone to the any of the FXS ports of the NanoPBX and dial “* *” to get the WAN IP
address of the NanoPBX. And then launch the web browser on the PC which is connected to the same
network to get GUI Login Screen.
Or also you can access the GUI of the NanoPBX by connecting a PC to the LAN port of the NanoPBX.
Enable the DHCP option in the Network Settings of the PC, and then enter http://192.168.113.1 in the
Web Browser Address field. (Where 192.168.113.1 is the default local IP address of the NanoPBX).
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On the login screen, the default username and password is “admin and admin” respectively. Press the

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Login button to enter the NanoPBX web panel. To change the password, please refer to the SETTINGS >
General section in the navigation.
Important Note:
Recommended to use Mozilla Firefox or Internet Explorer 6 & above version Web browser.
Setting up Features
3.1 System status
Navigation: > System status Status of nanopbx including Memory Status, VoIP Status, Networking Status
and Client Status (to see which clients are connected to the system).
After you login, you are brought to a System Status screen, which offers information about the NanoPBX,
and help files to assist you in learning about all the different features of the system.
•Under “Trunks” you will see the Registration Status of the VoIP Account(s) configured. If it displays
“Registered” then it is successfully configured and connected.
•Under “Conference Rooms” you will see all Conference Extensions available or unavailable.
•Under “Parking Lot” you will see all Parking Extensions available or unavailable.
•Under “Extensions” you will see all the users and the extensions connected to the NanoPBX. Also
you can sort out the extension by clicking in the extension
•Under “Queues” you will see all Queue Extensions available or unavailable.
•Under “System status > System Info > General” you will have a summary of your General
information, such as Hostname, Server Date & time zone, Uptime.
•Under “System status > System Info > Memory Status” you will see total memory resources,
including RAM usage, Compact Flash usage (to store voice files and voicemail) and Inbox status
(for voicemail inbox).
•Under “System status > System Info > Netwok Status” you will have a summary of your Network
information, such as Hostname, WAN IP Address, Subnet Mask, WAN MAC Address and Default
Gateway (you may refer to Settings > Network Settings for more info).
•Under “System status > System Info > Disk” you will have a summary of Disk usage and Disk free
space available on the filesystem.

3.2 Configure hardware
3.3 Trunks
Navigation: Trunks > Analog Trunks: Here you can configure the Analog Trunks.
3.3.1 Analog Trunks
The Trunks are configured or created by selecting the channels i.e. checking in the check box provided
next to channels which are grouped under Groups options.
The Trunk Details are as follows:
•Channels: select the channel which you want to use.
•Trunk Name: Name of the trunk which user wishes to use.
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•CallerID: Number to be displayed when the calls are routed through trunks.

Advanced options : These options are used to configure the various options which are user friendly such
as Busy Detection, Busy count, Busy pattern, Ring Timeout, Answering/Hanging based on polarity switch
etc. User while creating the trunks can create without setting the Advanced options as these options by
default are set. After all the changes are made click on update button followed by the Apply changes
button which is on the navigation bar at top right. User can even edit or delete trunks by using the Edit
and Delete button.
3.3.2 VOIP Trunks
Navigation: SETTINGS > VOIP Trunks: This is where you setup VoIP Trunk, or manage existing ones.
In this page, fill in the Provider Name, Host name, Host Port, Username, Password and Proxy information
given to you by your VoIP provider (known as SIP Credentials). Apply any Codec Settings required. You
can prioritize your active codec’s by using the drop down buttons.
After you have entered the details, click the Save button at the bottom.
Adding a New VoIP Account Details
•Provider Name: SIP service subscriber name to identify the trunk when listed in incoming and
outgoing rule.
•Host name: IP address provided by VoIP service provider.
•Host Port: Used to set the Proxy Port numbers.
•Username: User account information, provided by VoIP service provider (ITSP), usually has the
form of digit similar to phone number or actually a phone number.
•Password: Set a password to register to (SIP) servers provided by the ITSP.
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•Authorization User: Same as username.

•Caller ID: Caller id given here will be displayed to the other end for outgoing call using this VoIP
trunk
•Proxy IP Address: IP address or Domain name provided by VoIP service provider.
•From Domain: Domain name provided by VoIP service provider.
•Insecure: select the very option (recommended) which means includes all options.
Editing / Deleting an Existing VoIP Account:
On the right side of the page, you can see the list of VoIP trunk you have setup. To edit, or delete any of
them, simply click the appropriate icon provided to the right of each trunk.
Once you click on the edit button of a VoIP trunk then it will display the information of that particular VoIP
trunk, here you can change the required details and click on the Save button and then click on the apply
changes tab to save the changes made.
To ensure successful registration of your VoIP Trunk, you must click the “System Status” tab on the top
navigation menu (see Status section for more info)
Important Note:
Make sure to click the APPLY CHANGES tab in the top navigation bar, after adding any new VoIP trunk, or
editing / deleting.
3.4 Outgoing Calling Rule
Navigation: Outgoing Calling Rules > New Calling Rule: This is where you configure DialOut Rules.
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Outgoing Calling Rules represent the prefix sequence used to dial when making an outgoing call either

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through PSTN(Analog) or VoIP.
It is prefix used to configure and enable system to judge outgoing call via FXO or VOIP and also used to
select a Least cost Routing provider. This will allow user to configure add/delete of rule.
There are two ways to make outgoing calls for the registered extension users:
=> VoIP / SIP trunk via ITSP gateway
=> LINE / PSTN trunk via FXO port
Outgoing Calling Rules Details
Outgoing calls go through Trunk sequences. If it fails, it will go through the next selected sequence, and
so on. Please go through the Dial Pattern section to know about how to choose outgoing trunks.
•Calling Rule Name: Provide proper rule name
•Dial Pattern: A unique set of digits that will select this trunk. Enter one dial pattern per line.
A pattern consists of digits 0-9, digit set [digits], and wildcard characters like "." and "X". Below table
explains digit set and wildcard characters.
Expression Description
[digits]
Match any single digit listed explicitly. E.g., digit set [13579] match odd digits.
One may use
. (dot)
Is a wildcard, match any digit in any length i.e. one or more digits. Usually give
n
in the end
X Match any single digit from 0 to 9.
| Is a wildcard, matches any digit i.e. zero or more digits
You can even use the Calling rule to route it to local extensions as Destination or Route the calls to
Trunks created so as to make the calls successful. There as still many options like striping the number of
digits from front, prepend the digits before dialing and filter. There is also another alternative way if the
trunk fails to route the call i.e. Failover Trunk an alternative trunk to route calls.
When you have finished adding your Outgoing Calling Rules, click the Save button.
Here we will discuss about how to choose outgoing trunks in two different ways.
The First way is, choose a provider or a trunk based on prefix.
This type of rule will allow user to create a prefix for choosing SIP (VoIP) or LINE (PSTN / FXO) trunk to
make an outgoing call.
For e.g. adding '01|X.' in the Dial Pattern or entering 2 in field strip and selected a VoIP Provider1 in
Trunk Sequence, will allow user to dial 014567807890 numbers, so that 01 will be stripped off from the
dialed digits and only 4567807890 are dialed through VoIP Provider1. (Where as a VoIP Provider1 is an
account which is registered to any Internet Telephony Service Provider)
Second way is, choose provider or trunk based on actual number dialed
This type of rule will allow the user to choose suitable provider based on actual number. For e.g. adding '
44X.' in Dial Pattern and selected VoIP Provider 2 in trunk sequence, PBX will allow number dialed from 44

+ (any digit from 0 to 9) like 449872837532 will route through VoIP Provider 2. If the ITSP / VoIP
provider offer cheaper rates for the region where number starts from 44 users can make use of this rule.
In the same way, user can create a prefix and select a LINE (PSTN / FXO) trunk to make an outgoing call.
Navigation: Outgoing Calling Rules > Restore Default Calling Rule: This is where you configure Default
Outgoing Calling Rules.
These are the default calling rules where they show us an example of the patterns and which all fields can
be used as default.
3.5 Dialplans
Navigation: Dialplans: This is where you configure Dialplans for the users.
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A Dialplan is a collection of outgoing rules. Dialplans are assigned to users to specify the dialing
permissions they have
For Example: you might one Dialplan for local calling that permits the users of that DialPlan to dial local
numbers, via the “local” outgoing calling rule. Another user may be permitted to dial long distance
numbers, and so would have a DialPlan that includes both the “local” and “log distance” outgoing calling
rules.
You have to create the dialplan first before you create any user accounts to make your call successful. The
DialPlans details are as follows.
Dialplan Name: the name user wish to see in that field of Dialplan.
Include Outgoing Calling Rules: when the out going Calling rules are created it displays here so that to
include it in the dialplan.
Include local contents: here the user can select the features which he wishes to use.
After all the changes are made click on the save button. And don’t forget to click on apply changes button
on top navigation bar immediately after save button.
3.6 Users (User Extensions)
Extensions are the core of the NanoPBX. An extension is a number mapped to a person. So basically,
every employee that is connected to the NanoPBX should have their own unique extension number, so
that he/she can be reached, and be able to place calls.
The NanoPBX supports 2 types of Extensions: IP Extensions and Analog Extensions.
IP Extensions:
IP extension are devices that have only data networking connection such as Ethernet and they
communicate with the NanoPBX using IP based protocol for signaling and Voice, examples are IP Phone,
Soft Phone application. The NanoPBX can support up to 100 IP Extensions registration.
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Analog Extensions:
An Analog Extension is used with a regular telephone system which can be connected to an available FXS
port on the back of the NanoPBX. The NanoPBX can have up to 6 Analog Extensions.
All the features are supported by both the Analog and IP extensions. Extensions can be part of other
features, such as: Queues and Voicemail Groups. Also, the extension can have a voicemail of its own.
3.6.1 Create New User:
You have to create atleast one Dialplan using Dialplan option before trying to create Ananlog/IP
Extensions
Navigation: Users > Create New user: This is where you setup your Analog/IP extensions
Analog/IP Extensions:
To add a new Analog/IP Extension, First select the Dialplan option by choosing from one of the available
Dialplans. Then, fill in the required pieces of information, such as Extension, CallerID Name, CallerID
Number and select if you’d like to make that extension available for Voicemail configuration. You have to
select the technology (See to that Check Box next to SIP Technology in not marked) from the drop down
box which displays the ports available for creating the analog User Extension. You have to checkin the
Check box next to SIP Technology for creating the IP Extensions. You can also configure various Codecs
and Other Options available for using users in other features. Once you are done, click the Update button.
Details are given Below
Extension : Unique identifier (i.e. :6000)
CallerID Name : Display name of the user for the called user
Dialplan : Select one from the dropdown box (appears only if dialplans are created using Dialplans)
CallerID Number : Is same as Extension field it’s a number which displays to called users
Enable Voicemail for the User : Enable this extension so that it can be visible when adding members to
the Voicemail Group and Voicemail. The Voicemail PIN password will be same as the password provided to
the Extension while creating an Extension. This will applies to both Analog and IP Extension. If you are
trying to access or retrieve Voice Message, use that particular Extension password.
Email Address : Mail address which the user wishes to receive mails(for Voicemail)
Technology :
a) For analog user Extension: select the ports from the drop down box. It disables the VOIP settings fields
as they are reserved only for VOIP extensions

b) For IP user Extension : mark in check box next to SIP
Codec Preference : select the codecs from the drop down boxes
NAT : for IP extensions enable the NAT by marking in the check box
Insecure : select very (recommended) its optional
Other Options : select them as they makes user extensions support for other features.
The Other fields are optional they will be set to their default values when created.
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3.6.2 Modify/Delete selected users:
Navigation: Users: This is where you can edit / delete existing Analog/IP Extensions individually.
On the right side of the page, you can see the list of extensions you have setup. To edit, or delete any of
them, simply click the appropriate icon provided to the right of each account/Extension.
Once you click on the Edit button of an Extension then it will display the information of that particular
extension. Here you can change the required details and then click on the Update Extension button to
save the changes made.
You can delete an extension by clicking on the delete button of the extension from the list of extensions
displayed.
Navigation: Users > Modify Selected users: This is where you can edit / delete many existing Analog/IP
Extensions simultaneously.
On the left side of the page, you can see checkbox with the list of extensions you have setup. To edit, or
delete any of them, simply mark the appropriate check boxes provided to the left of each
account/Extension.
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Once you check the boxes click on the Modify selected users tab. Here you can change the required
details and then click on the Update button to save the changes made.

You can delete many existing extensions by clicking on the Delete Selected Users button after marking in
the check boxes of extensions from the list of extensions displayed. Click ok the popup window to delete
the selected users.
Important Note:
Make sure to click the APPLY CHANGES button in the top navigation bar, after adding / editing / deleting
any Extension. The APPLY CHANGES tab appears if some changes are made and not saved.
3.7 Ring Groups
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Navigation: Ring Groups: This is where you can create/edit /delete existing Ring group Extensions.

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Ring Groups distributes incoming calls in the order of arrival to the first available extension in the list of
available users based on the strategy. The system answers each call immediately and, if necessary, holds
it in a wait until it can be directed to the next available extension after seconds mentioned to ring each
member.
This feature is used to balance the workload among group of extensions.
Ring Groups will provide the following functions,
Incoming calls being routed to extensions based on the strategy.
Extensions that are available for the call in the Ring groups.
A strategy for how to handle the calls in the queue and distribute the calls in the queue.
Configuring a Ring Groups
To create a Ring Groups, simply fill in all the required details including Extension, Ring group Name,
Strategy, seconds to ring each member and select the other options. When done, click the save button.
Ring Group Details
•Extension: Extension number to reach the Ring Group directly
•Ring Group Name: The name of the Ring Group.
•Ring Group members: This is the list of available extensions that could be part of this Ring Group
by moving them to right box.
•Available Members: This is the list of extensions that are members of this Ring Group.
•Strategy Type:
Ring in order: Takes turns ringing each available extension in sequential order.
Ring All Simultaneously: Rings all available extensions simultaneously.
•Seconds to ring each member: Select the time required in seconds so that the Rings of each
extensions in Ring Group will get terminated after the given seconds
•If not answered Goto: If none of the members answered the calls this is field where we specify the
cause for the Ring Group call.
•Ignore redirections: If we enable this we can ignore redirection of the Ring Group Call.