The Mizu PBX is a Class5 softswitch application running as a service on the
Microsoft Windows operating systems.
Modules: SIP, WebRTC, RTMP,
access roles, routing (priority, weight, BRS or LCR), billing, PBX, accounting,
CDR records, blacklist/whitelist filtering, IVR, transcoding, call recording, voicemail,
conferencing, conference rooms, media server, alerting, statistics generation,
client applications, API and others.
The PBX features and easy to
use installer and configuration wizard: download windows pbx from here.
MizuManage
All
administration and monitoring tasks can be done from the MizuManage (MManage
/ MizuManagement) admin client which is included in the install package.
Nearly all
forms can be filtered with the following filters:
·
Quick filter: found in the top-left side in
MizuManage. For example type “44*” in the quick filter box then open the “CDR
Records” form and click “Load”. You should be able to see all calls to 44…..
numbers. Or enter “test” and open the “Users and devices” form. Click on the
load button to see accounts containing the “test” word (in name, username,
address, etc)
·
Direction
filtering:
accessible by double clicking on the space above the quick filter or from the
Settings menu -> Set direction. When you are doing operation which needs
more precision (eg. billing), always use the Set Directions form and not the
quick filter.
·
Date-Time filter: found in the top-left side in the
MizuManage. Useful to restrict statistics, reports and CDR listing intervals.
To export
data from the application, use File menu -> “Save As”. A more advanced
export tool can be found in the Advanced version, accessible from File menu
-> “Export/Import”.
From the “Edit” menu you can manipulate the selected dataset (grids, etc).
The Config menu might be used only one time during the initial setup. Under the
“Utilities” section you can find many tools to perform various operations like
sending sms, playing recorded voice, etc.
Quick
Tip: you can filter almost any forms using the quick search and the date-time
filter.
Quick Tip: always check the status bar (the bottom text display). MManage
rarely displays popups and the success/error status of various operations is
displayed on the status bar instead.
For the basic server configuration you
should walk through the configuration wizard accessible from the Config menu.
Don’t change any setting that you don’t fully understand, just click on
the “Next” button in this case. Most of the settings are self-explanatory with
a short description near them and a hint if you hold the mouse over a control.
Take special attention for the IP and bind IP settings if your server
has multiple IP’s assigned.
Take special attention for the NAT and Public IP setting if your server
has multiple IP’s assigned or your server is behind NAT.
After you have finished with the configuration wizard you might have to continue
with the following tasks:
- add your outbound routes and traffic senders: Access -> Users and devices
-> Sip Proxy and Traffic Senders, Routing
- add users: Access -> Users and devices -> Enduser
- fine-tune other settings: billing, blacklists, etc.
For more advanced options you will have
to change global config options manually on the “Configurations” form. (Under
“Other”).
If you are
not sure where to find a specific configuration option, search for your keyword
in the “Configurations” form and also in the admin guide.
Quick
Tip: right click on the “Configurations” node in the main tree view to have
some grouped options.
Once you
have gone through the configuration wizard, you can start the service (From
Control menu -> Start server or from Windows Service manager locate the
“mserver” entry and right click to start). Once the service was started open
the “Analyze” form to check for any potential issues.
The server
will allow calls between endusers by default, so you can already test by
registering with 2 softphone (such as X-Lite) and make calls between them. See
the first test calls section for more details.
Listing users
Users can represent real people, extensions,
devices or virtual endpoints.
Open the “Users and Devices” form
(below the “Access” section) and (double) click on Endusers.
You can apply various filtering using
the user “Type” checkbox-list, the dropdown-list on the top of the form or the
already discussed direction filter or quick filter.
Quick Tip: You can easily
search for users using the “quick filter” box. For example to list all outbound
routes whose username or name contains the “carrier” word, select the “SIP
Server” and type “carrier” in the quick filter. The quick filter will also
search in other fields such as IP, name, email and others.
Quick Tip: you can list
user from Users and devices form by right clicking on the tree view user type
node.
Quick Tip: There are many
fields in the user table which can be used for various settings. To quickly
find a field, double click on a user node (such as “Endusers”) then click on
the “Search fields” below the grid. Another way is to use the “Show Fields”
from the “Fields” menu
The best way to create new users is to
clone an existing working account with the same user type.
For this, launch the “Users and
Devices” form, select a user type, and click on the “Load” button. Then
select any user entry and click on the “New User” button.
Endusers are the most commonly used type (retail
customers/company employees). Usually you will select “Username/password”
authorization for this type of users and enter a valid username and password. The
username can be also used as a real phone number; will be used also as the
caller-id if not specified otherwise. Endusers can make voice or video calls
between them for free of charge and also IM and presence are enabled by
default. By default the server will route the RTP if needed (if users are
behind NAT) or allow it to bypass your server saving your bandwidth. Sub-Endusers
can represent VoIP devices, extension, child accounts or callshop cabins. Calls
from sub-endusers account are billed for the parent Enduser.
Traffic sender users are used for receiving traffic from other SIP
servers and carriers. The authorization type is usually set to “Auth ip must
match” and you have to enter a correct “Auth Ip” (IP address based
authentication). If you don’t have special requirements, the only thing that
you have to communicate to your peers to be able to send calls to your server
is only your IP address. (Your server needs a public IP for this or you have to
setup proper port forwarding).
For outbound traffic you need one (or
more) SIP Server user. The most important parameter here is the “IP”
where the VoIP calls will be sent. Then you will have configure these also on
the Routing form. To be able to send and receive traffic to/from another SIP
server or carrier you will have to add it as both a “traffic sender” and “sip
server” user.
To import endusers from other
data sources, use the Config menu -> Users.
You can also generate users in bulk
from the “Generate users and PIN’s” menu item.
Users can also sign-up to your service
with the following methods:
-From the enduser web portal (“New user”
link on the login page)
-From customized softphones (all of them
have a user interface where new users can sign-up to your service)
-Integrate the sign up capability into
your web site or any application using the Mizu PBX newuser/adduser API
By default the new users can already
call each other for free (unless you set otherwise). If you use the server for
your own company, then you can create new user record as postpaid so they will
be able to call outside by default. Otherwise the user records are created as prepaid
and must have positive credit balance to be able to initiate outside calls.
Quick Tip: Right click on
the “New” button for other options to add users. Endusers can be added more
comfortably from “Add Enduser”.
Unless in
other traditional PBX’s, in the Mizu PBX you can just add a new user with the
“username” and “password” settings to be reachable for incoming calls also.
Then the “username” field (which can be a phone number) will act as a
SIP username for authentication but also as an extension number or a DID
number. You can also use the same username/password to login on the enduser
web-portal and in any other operations requiring authentication. If you wish to
use a separate username and CallerID for the users, then you can enter the
Caller ID to the “Other numbers” edit box. If one user has multiple DID
number assigned, then you can add then using the “…” button near the “Other
numbers” edit box (Users and devices form, Edit tab). More than one users can
share the same DID number: simply add it to the required users as “other
number” with type 0. The call will be routed to the “best” device (based on the
user status whether it is registered or in-call). You can also setup ring-groups
to allow forked calls to ring on multiple devices.
Quick
Tip: you can get real phone numbers for your endusers from providers like DIDx or from your carrier/sip trunk provider.
This is needed if you would like to
route the calls from the users to another servers, gateways or carriers (for
example calls to mobile and landline numbers).
Add a “SIP server” user from the
“Users and devices” form with any meaningful username and make sure to set the
IP field correctly on the Edit tab. (This field can also contain a domain name
instead of IP address). Some carriers require the usage of a techprefix which
can be entered in the “Tech Prefix” field. Others settings are rarely used.
You must also add this server(s) to your
routing:
Open the “Routing” form. In the
left side you have to define your pattern which will restrict the condition
when the actual route entries can be used. If all fields are empty and the time
definition is set to “All times” then all patterns will match. You can make
restriction if you make specification here (caller, called prefix, time
restriction, etc) . Make sure that you increase the priority for the pattern
(to be higher than the your “general” pattern where you have not made any
restrictions)
On the right side you will have to add
one or more sip proxy user. If you set more than one route with equal priority,
then you have load balancing, LCR or BRS (depending on the “brs_lcr” global
config option); otherwise the traffic will be routed after the prioritizations
(will flow to the lower priority servers only if you have reached the maximum
port limitations or because automatic failowering). For more details please
read the routing guide.
Usually for
a B2B usage, uppers servers (your carrier or VoIP service provider) will setup
IP based authentication. This is the favorized method for a trunk
interconnection. If your outbound server (where you are sending traffic and
receive incoming calls) needs username/password based digest authentication
instead of IP based authentication, you can set it from the “SIP server” user
configuration. Create a SIP server user, then switch to the “Edit” page.
On the bottom of the page you can find a grid named “Proxy authentication”.
Here you can add the login details (multiple username/passwords can be used for
your convenience). Then select “Username/password must match” from the
“Authorization” drop-down list.
These are the basic and
most commonly used authentication settings. There are many other combinations,
for example you can forward the username/password as received from your users.
For more details please consult the Admin Guide.
If you would like to accept traffic from
other servers (for example you are doing a wholesale business or to offer VoIP
trunk services for other companies), then you have to create “Traffic sender”
user. Usually you can use IP based authentication. For this, add the peer IP to
the “Auth IP” field.
For each incoming calls, the server will
first check if the called party is a local user. If not, than the call are
routed regarding the rules which is set by the “Routing” form.
Actually you could also use “Enduser”
users for the same thing, but for a bigger traffic volume it is always to
differentiate normal endusers from “traffic sender” so your statistics will be
easier to understood.
Quick Tip: You can easily
create a transit server by using only “Traffic sender” and “SIP server” users.
Quick Tip: One “Traffic
sender” user entry can handle incoming traffic with IP authentication from
multiple IP addresses . Use the “…” button near the “Auth IP List” to add more
IP address.
First test calls
For a test
call create 2 enduser accounts with username/password authentication.
To find out how to connect to your server see the “Client configuration” from
the Help menu.
Register with two softphones and call from the first account
to the second account.
Softphone configuration:
·
domain: your server IP or domain name (and the server port if
your server is not using the standard 5060 UDP port)
·
proxy: you can leave it empty
·
username: the “username” field from the newly created user
(tb_users.username)
·
password: the “password” field from the newly created user
(tb_users.username)
No any other special settings are required (such as
NAT, STUN, etc).
The network setting should be automatically handled by the
server. If you don’t hear any voice you might change the RTP routing for the
user(s) to “always route RTP” from MizuManage -> Users and devices ->
Edit tab.
During the call, you can open the “Current
calls” form in the MizuManage to see the details.
After the call you can see the CDR by
opening the “CDR records” form in the MizuManage.
If there are no CDR records, it means
that the call has not reached the server (wrong network settings on server or
client side)
Troubleshooting:
If you can’t register or make calls,
check the followings:
·
Have a look at the “Dashboard”
·
Have a look at the “Analyze” for
to detect any potential issues
·
Check any important errors or
warnings on the “Logs” form
·
If you are making call to a local
user, make sure that the called user is registered when you call it
·
In case of call failure you can
check the disconnect reason from the “CDR records” form
·
Verify the log files. You can find
the logs in the server app folder (near the mserver.exe) -> “logs”
subfolder. You can also open the folder from MManage -> Files Menu -> Folders
-> Server logs.
·
Open the last logfile (“log_xxx.dat”)
with a fast text viewer (For example F3 from TotalCommander). To find
application errors, open the last log file in the server app directory and
search for “ERROR” and “WARNING”. To find a call, search for “INVITE
sip:callednumber”.
·
You can modify the trace details
with the “loglevel” configuration option: from 1 to 5.
Setup billing [optional]
This is necessary only if you wish to
track your spends or to limit/track employee call costs.
User to user calls are not charged by
default (this can be changed with the “internal_endusercost” global
configuration option).
All other calls are checked against the
user credit and prepaid/postpaid setting which can be set from Users and
devices form (select a user and switch to the “Billing” tab)
From here you can assign a billing
packet for the user explicitly but the better way is to setup one or more
packets to be valid for all your users, for a group of users or on special
circumstances (caller, called, techprefix, time, etc)
These packets (call rating) can be set
on the “Price setup” form (below the “Billing” section)
·
On the left column add a billing
group with any name (“default” is ok). This is used only to logically group
your tariffs but not used by the server.
·
On the middle column specify your
conditions. You should have at least one Enduser cost type without any further
restriction on the traffic direction (so it will be applied for all endusers/directions/time)
·
On the right column enter or
import your pricelist applied in the conditions defined by the middle column.
For a default price enter prefix “*”
(this will be applied to all destinations that is not specified explicitly)
Make sure you have set the proper
currency (in the global configuration, in the price setup and also for your
users)
Read the Billing guide
for more details.
The Mizu PBX provides
endless possibilities for monitoring both real-time and statistics. Some of the
most important tools are the followings:
Dashboard: a summary of the most important parameters and a
start point for management
List the active
sessions: Monitoring -> “Current
Calls” form
Call detail record: Monitoring -> CDR Records form
Statistics by users: Monitoring -> Statistics -> Group By: caller
Statistics by day: Monitoring -> Statistics -> Group By: day
By using the “Analyze”
form you can have a quick overview about the system.
Other more advanced statistics
can be generated by using the Statistics form and using different
fields/options/grouping/directions.
All statistics can be filtered by the
“set direction” form or the “quick filer” edit-box and by a time interval
selection.
Automatic reports
The server can send daily reports for
administrators or email/sms alerts on malfunctions. For this you have to setup
an “Admin” user with a valid email address. Then set the following fields to 1
(after your needs): “sendemailalert”, “senddailyemal”, “sendmonthlyemail”,
“sendsmsreport”, “sendsmsalert”
The server will be able to send SMS
messages only if an SMS provider is configured (see the Admin Guide)
Quick Tip: You can access
various statistics by just clicking on the Dashboard items. For example click
to “CCalls” will show the current calls.
CDR Records
CDR records can be listed by using the “CDR
records” form. By default only the most important fields are listed
(date-time, connect time, call duration, etc). You can see more details if you
check the “All fields” checkbox.
To quickly list the CDR records that
belongs to a user, open the “Users and devices” form. Find the user record,
then right-click on it and select “Set Direction”. Than go back to the CDR
record form and click on the “Load/Reload” button.
If you have enabled voice recording
for some users, than you can play the recorded audio by filtering for “Recorded
Conversation” (select the desired record and click on the Play button)
Quick Tip: You can easily
filter calls with a specific prefix by typing the prefix in the quick search
box following with an asterisk and hit enter. For example searching for 44*
will list all CDR records where the called number begins with 44.
Dial plans
You can manipulate number format, SIP
headers or the Caller-ID from the following settings:
·
Users and devices form: caller-id,
username, other numbers (DID match), tech prefix
·
Routing form: you can only specify
routing direction here without number changes
·
Rules form: this is a powerful module which you can use to
change almost anything (including caller-id, called number format and many
others)
·
Global configuration: a few global
configuration options might also affect the dial plan
·
Prefix rules and the dial plan
form: use the “Rules” form instead when possible.
You can find more details in the VoIP Admin Guide below the Routing section.
PBX features
A number of PBX features are enabled by
default. More PBX modules can be activated if you select the “PBX Extra” on the
configuration wizard or set the “fs_pbx” global configuration options to “1”.
This includes:
·
call forward: can be enabled from users and devices form ->
functions tab
·
call transfer: available as specified SIP standards (via all
devices with support for transfer)
·
voicemail: enable from users and devices form -> functions
tab. Default access number are 5000 (with pin) and 5001 (with auto
authentication). Auto-email forward is enabled by default.
·
conference: via SIP standard, via dtmf *1*number# or via
conference rooms using extensions between 5100-5199 for narrowband and
5200-5299 for wideband
·
call recording: just check the “Voice Record” on the users and
devices form -> functions tab. Playback the recorded voices from the “CDR Records”
form.
·
special numbers and IVR’s such as music, record/playback, vide
record/playback and others
·
missed call notifications by email
·
barge in: via the “Voice here” form, or right click on current
call or 5009 access number
·
many others such as caller-id, ring groups, call hold, call
waiting/park/pickup and others
You can setup your calling card or
callback business by using access numbers and assigning them to one of the
existing or newly created IVR’s. You should be able to request DID numbers
from your existing VoIP carrier or by contacting other companies e.g. www.didx.net. In this case you will have to add it as a Traffic
Sender user usually with IP based authentication (fill the AuthIP box with the
provider IP or domain name)
After you have terminated with the
traffic sender configuration, you can add the access numbers like usual
endusers. Type the phone number in the “username” field or you can also use the
“SIP number” field for the same reason. Then switch to the “Functions” tab and
set the “Campaign ID” and the “Callback access” (if the DID number will be used
as a callback access number); optionally you can enable A number authentication
(PIN less dialing). The campaign id means the ID field from the tb_ccampaigns
table (You can see them by opening the “Campaigns” form).
For more complex authentication and
billing options please consult the admin guide.
IVR
setup [optional]
The IVR module is used for various tasks
like access numbers, calling-card operation, customer support etc.
You can assign different IVR’s to
different access numbers by using the “Campaigns” form. To create a new
campaign, just click on the + sign and enter a “name” for the new record. The
most important configuration for an IVR campaign is the script. Switch to the
“details” tab to select a “Script”.
Scripts can be created by using the “IVR” form. The server is
shipped with several preconfigured script examples, but you should easily add
new scripts or modify the existing ones by following the admin guide
or the IVR documentation.
Quick Tip: you can also
offer callback services via the IVR module with a script which will reject the
incoming calls and call back the caller automatically.
Provide SMS services to your users by
interconnecting with an SMS provider capable to provide a HTTP API. For this
just set the smsurl global config option or create SMS GW endpoints if you wish
to use more than one provider. A guide can be found here.
To setup an outbound sms routing,
you have to contact a company providing SMS services. (For example Clickatel)
Then open the “Configurations” form and
search for “smsurl”. Enter the details in this format:
http://api.clickatell.com/http/sendmsg?api_id=APID&user=USERNAME&password=PASSWORD&to=[tonum]&text=[message]
Pricing is done after the “smsprice”
global config options or you can setup detailed pricing by using the “price
setup” form.
Users will be able to send sms messages
by using a softphone, the webportal or there is a possibility to create an SMS
sender application yourself by using the http or database api.
For incoming SMS applications (SMS
callback, balance request, etc) you will have to request a two way
SMS service (to get a DID number)
Mizutech provides customized VoIP
clients for web, desktop and mobile as part of the all in one softphone package or separately, so you can offer preconfigured dialers for your
users with your name, logo, icon, branding, links and many options:
·
Browser webphone
·
Windows softphone
·
Android softphone
·
iOS softphone
·
Symbian dialer
·
Other softphones
You can also use your IP phone, third
part softphone (such as X-Lite) or any SIP compatible device.
Compatibility with browser VoIP clients
is assured by the built-in WebRTC and RTMP modules.
Others
·
Easily and transparently convert
between various VoIP protocols by just enabling them on the config wizard:
SIP/H.323/WebRTC/RTMP
·
The VoIP service is based on open
protocols so you can integrate it
in any environment.
·
Provide call termination / SIP
trunk services for other companies. Just set them as Traffic Senders (specify
auth IP), send your server address (IP or domain) to these companies, set
proper pricing and you are ready to go
·
Codec transcoding by setting the
“needcodecconversion” and “choosecodecs” fields for the user records
·
You can easily backup all data by
just copying the mserver.sdf file
·
You can enable many other
features
Common terms
·
SIP: The Session Initiation
Protocol (SIP) the most important signaling protocol used for VoIP
·
WebRTC: is a media streaming
protocol found in modern browsers
·
RTMP: is a media streaming
protocol used by Flash
·
RTP: media channel protocol (used
for audio/video/fax routing)
·
SIP Trunk: in-bound or out-bound
links. In MizuManage this can be set as “SIP Server” and “Traffic sender” users
·
Global Configuration:
configurations applied to all users stored in tb_settings. Changes can be made
from the “Configurations” form
·
User Configuration: configurations
specific to a user. Changes can be made from the “Users and Devices” form
·
Dial Plan: describe the format of
the phone number. In MizuManage this can be changed by global configuration
options, rules, prefix rules and from the “dial plan” forms.
·
PBX: means “Private Branch
Exchange”, which is a phone network used by companies. Users can call and chat
with each other and share a number of outside lines (DID) for external calls.
·
Callback: DID or toll free number
configured as enduser with iscallback set to the required IVR
·
Click-to-call: a html button
placed on websites to request an immediate connection with another person by
phone call. Can be implemented using the webphone
·
ASR: Answer Seizure Ratio. The
percent of the connected calls compared to all calls (How many calls are
connected from 100 calls in average)
·
ACD or ACL: Average Call Duration
(measured in seconds for connected calls only)
·
IVR: Interactive Voice Recognition
(used for calling-card for example)
·
ANI / CLI: Automatic Number
Identification or Caller Line Identification (important on IVR systems for user
authentication)
·
ANI callback: same as callback
with User-ID based authorization (A number)
·
LCR: least cost routing (routing
the traffic to the least cost outbound server)
·
BRS: best route selection (intelligent
routing based on price + quality + other statistics and settings)
Resources
·
For more details, please consult
the Softswitch admin guide and other server related documentations on our website.
·
You might purchase a support plan
as described on the “Support plans for Windows PBX” section on the Support Terms
page.
·
Contact serversupport@mizu-voip.com for more help.
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