Requirements.

Voipswitch system is compatible with Windows 2000, Windows XP and Windows 2003.
SQL database is also required, either MS SQL or MYSQL version 4. MYODBC driver is
recommended to connect MYSQL database with VoipSwitch application. It can be
downloaded from the www.mysql.com website.
In addition it is recommended that MDAC version 2.8 or higher be downloaded (more
information available on www.microsoft.com)

Program installation.

After extraction files from VoipSwitch.zip three catalogs will be created: “VoipSwitch”,
“hearlink” and “callback”. In order to install VoipSwitch, it has to be saved to the folder
that has the same name and is located on the server; and installation program has to be
launched. Installation program will create VoipSwitch group in the “programs” menu.
The next step is to create a connection with the database. In order to accomplish that
VPSConfig program has to be activated. After running VPSConfig, “database settings”
window will open and “create connection string” button has to be chosen. After clicking
on “create connection string” button, “data link properties” window will appear with the
list of database drivers. Depending on the type of database that is used the following steps
have to be completed:

MSSQL:
1. From the list of drivers “Microsoft OLEDB provider for SQL servers” has to be
chosen.
2. “Next” button has to be clicked on.
3. Name of the server or IP address has to be entered or chosen. Depending on the
way user logs in to the SQL, button “Use Windows NT integrated security” or
“Use specific user name and password” has to be chosen. In case of the
subsequent all the necessary data have to be entered and “allow saving password”
button chosen. “Test connection” button can be used to check if the connection
succeeded.
4. If the connection succeeded, “OK” button has to be clicked on which will result
in closing the “data link properties” window.
5. “Create database structure” button has to be chosen. From the unfolding options
with the database types MSSQL has to be chosen and confirmed. After
completing all the above steps the information confirming successful creation of
VoipSwitch database should appear.

MySQL:
1. From the list of drivers “Microsoft OLEDB provider for ODBC drivers” has to
be chosen.
 2. “Next” button has to be clicked on.
3. “Use data source name” option has to be marked and the name representing the
connection to MySQL has to be chosen. During the installation of MyODBC the
connection named “myodbc3-test” is created by default. After choosing it
database’s username and password have to be entered, unless they have been
specified before. “Test connection” button can be used to check if the
connection succeeded
4. If the connection succeeded, “O K” button has to be clicked on which will result
in closing the “data link properties” window.
5. “Create database structure” button has to be chosen. From the unfolding options
with the database types MySQL has to be chosen and confirmed. After
completing all the above steps the information confirming successful creating of
Voipswitch database should appear.

After the proper creation of the database structure and connecting with it, menu on the
left side of VPSConfig window will become active.

Picture 1 Connecting to database

Initial configuration of VoipSwitch system.

“VoipSwitch settings” menu.

Call settings
“limit ring time” – activating this option allows setting up the maximum time
(measured in seconds) after which attempt to connect a client with a desired
number is terminated.
“limit call duration” - activation of this option allows setting up the maximum
connection time, after which the connection is terminated.

Authorization
“Authorize incoming calls” – this option is set as default. Switching it off will
result in accepting all incoming calls regardless which user they originated
from – in this case billing is switched off.
“Users can log by IP number” - this option is set as default. This option allows
authorizing incoming calls base on the IP address they originated from.
“Authorize by ANI” – this option enables authorizing incoming calls base on
the calling party number i.e. the number of the device (telephone) the call
originated from.

“Allow pc phone users log as Callback users” – activating this option allows users to use
one account both for pc2phone and callback services. After setting up account in
“pc2phone clients” menu user can also use the callback service using the same pair of
login and password.

“H323 settings” menu

The available IP addresses have to be chosen in this window, which listener h323 and
Gatekeeper will operate on. More then one address can be chosen here. It is also possible
to change the ports’ numbers, which by default are set on 1720 for h323 listener and 1719
for Gatekeeper. As for Gatekeeper the name has to be set and entered in the “Gatekeeper
ID” field. Also, “time to live” (by default set on 600 sec.) has to be set i.e. time after
which gatekeeper will regard the client as being logged out (provided the client will not,
in advance, send an information confirming his status as being active.
“Authorization”
“User can log by h323id” – with this option activated VoipSwich authorizes incoming
calls base on the h323id sent by a client. This is particularly useful in case of authorizing
clients who initiate connections from the dynamic IP addresses. After selecting this
option separating character has to be set i.e. character that separates login and password
sent as one string representing client’s h323ID. By default “@” acts as a separating
character. Thus, devices of h323 type should use “login@password” string as their
h323ID in order to be authorized. Also, separating character has to be set for the clients
who register into Gatekeeper.

“Sip settings” “pc2phone settings” “callback settings” menu

SIP Protocol’s settings are limited to choosing IP address that registrar and SIP listener
will operate on, and choosing ports (by default 5060). Also, “time to live” for the SIP
registrar has to be set. The procedure described above also applies to “pc2phone settings”
and “callback settings”

Launching the main application “VoipSwitch manager”

After the initial set up of VoipSwitch parameters the main application “VoipSwitch
manager” can be launched. It is installed in the menu programs/Voipswitch/Voipswitch.
After starting the application the information on the active services, previously defined in
the VPSConfig application, should be displayed in any of the windows showing the
applications’ logs.

Adding carrier’s gateways and gatekeepers.

Voipswitch is a system that receives traffic from clients and sends it to the VOIP
gateways or gatekeepers. As clients can also include VOIP gateways, Config is divided
into two menu groups. The first represents clients. Device’s accounts, or software
application are added here and the traffic is directed to switch. The second is the
“gateways” and “gatekeepers” menu where devices, which are responsible for accepting
calls sent from switch, are defined.

Gateways

Term “gateway” may refer either to the big carrier systems handling international traffic,
small gateways, Ipphone, or even software that is able to receive calls. In order to add a
new gateway user has to access the “gateways” menu, located in Config, and add a new
entry representing the new gateway. It is necessary to enter IP address, port (by default it
is 1720), codec that is used by the gateway and also to define whether the connection is to
be established in the FastStart mode or not. The name of the gateway entered in the
“gateway description” field is established only for management purposes. The name
entered in the “H323 ID” field describes VoipSwitch’s h323id that is sent to the given
target gateway.
For any given gateway two codecs can be chosen: g723.1 and g729. If the gateway can
operate on both then one has to be set as a primary and the other as a secondary. Having
only these two codecs as an option does not exclude others. This, however, requires
different proxy set then “full proxy” (for more information go to “dialing plan” section).
Switch can operate both with h323 and SIP protocols. Thus, while defining the gateway
either “h323 device” or “SIP device” has to be chosen.
Note: switch undergoes h323->SIP conversion and SIP->H323 e.g. SIP clients can
connect with the h323 gateways.
Picture 2 "Gateways" menu

Gatekeepers

Gatekeepers that VoipSwitch will register and then send the traffic to are defined in this
menu. Important: VoipSwith registers to the external gatekeepers as a gateway (not as a
gatekeeper). In order to properly register, data have to be entered in the following fields:
• IP address
• Port – by default it is 1719
• GK name – external gatekeeper’s ID
• Password – the password sent by VoipSwitch to gatekeeper. Required
only with the h245 registration.
• H323ID – VoipSwitch’s h323ID sent to the external gatekeeper in the
process of registration.
• E164ID – VoipSwitch’s e164 alias sent to the external gatekeeper in the
process of registration.
• Time to Live – established by VoipSwitch in the registration process
• Description – the name related to a particular gatekeeper, used only for the
convenient management of VoipSwitch. This name is not sent to the
external gatekeeper.
 • Active – if marked, VoipSwitch will try to register to the particular
gatekeeper.
“Connection properties”
• Primary, secondary codec – in these combo lists, codecs that will be
applied to the traffic going from VoipSwitch to the particular gatekeeper,
have to be chosen.
• “FastStart” – if marked all the connections from VoipSwitch to gatekeeper
will be conducted in the faststrat mode.

After adding a new gatekeeper the “relog to GK’s” button has to be clicked on. It is
located in “Voipswitch Manager” interface and clicking on it will result in an attempt
made by VoipSwitch to register to the particular gatekeeper.
If the registration process was successful there should be a blue icon displayed in the
right lower corner with the gatekeeper’s description. If the registration failed there will be
a red icon displayed in the “logs” window with the error description.

Picture 3 "Gatekeepers" menu

Creating dialing plan

Dialing plan allows defining routing i.e. describing where to and how to direct the traffic
from the clients to the particular phone number. Unlimited number of records can be
added to dialing plan menu, creating the complex routing structure that enables directing
incoming telephone traffic to various gateways and gatekeepers.

Picture 4 Creating dialing plan

For instance, to define routing for the connections with Poland (prefix 48) the entry may
include the following elements:
“telephone number” – 48, (country code)
“priority” – 0
“external gateway” – gateway 1
“proxy properties”/”proxy all connect independently” – active
“tech prefix” – 888
“dial as” – 48
“media wait for connect” – inactive
“IE Display from client” – active

Above configuration will result in all the incoming traffic coming with the prefix 48 to be
directed to Gateway 1 (defined previously in “gateways” menu). Connections will be
carried out in the “proxy all, connect independently” mode (proxy classes will be
described in the following sections). Prefix 888 will be added to the incoming number
and in such format will be transferred to the final gateway e.g. 48600316058 will be
replaced with 88848600316058. ID display (caller ID) will be transferred to final
gateway. Option “media wait for connect”, if active, enforce client’s device to generate
fake ring tone. If the final gateway sends real ring tone this option should be inactive.
In the above description it was assumed that client’s telephone number was written in the
e164 format i.e. country code + area code + number. Apart from this format it is also
possible that a client will send phone number written in a different format e.g.
0048600316058, or with the particular prefix e.g. 3334860006058. Since final gateway
only accepts calls to the 88848…numbers, it is necessary to use the “dial as” to make
above mentioned variations operational. Modification could look as follows:
“telephone number” – 0048
“dial as” – 48
“tech prefix” – 888
Above configuration will first cut “00” in the “00486006058” number and then to
remaining “486006058” 888 prefix will be added. As a result the number reaching the
final gateway will have the following form “888486006058”

It is frequent that VOIP provider has many gateways or cooperates with many carriers in
order to provide that highest quality of service. Apart from the main route to the
particular prefix, as described in the Gateway 1 example, provider also has other
gateways or gatekeepers, which carry out connections to the same destination. In order to
secure additional route to the particular destination the entry with the same “telephone
number” has to be added, for instance, 48 with priority set to 1. Afterwards, suitable
gateway has to be chosen e.g. gateway 2, with the appropriate parameters, such as “type
proxy” or “tech prefix”, defined. Defining the second entry referring to the same prefix
will result in VoipSwitch trying to send the connection to Poland through the gateway 1
and if this attempt fails (e.g. gateway is shut down) will automatically redirect the
connection to the gateway 2.
Using above described method many backup routes can be added.

In the example analyzed before only one entry was created for the 48 prefix – whole
Poland, without distinguishing between area codes or mobiles. This type of entry is
rational only if a particular gateway: Gateway 1 or Gateway 2 carries out all the
connections within the 48 prefix. ("dialing plan" settings have no effect on clients’
billing, which is entirely based on codes (prefixes) in a tariff).
If there is a separate gateway that carries out connections exclusively to e.g. cell phones,
and because of the quality concerns or prices it is a priority that the gateway would carry
them out first, it is necessary to add a new entry to “dialing plan” with more detailed
description of the prefix in the “telephone number” field. In case of Poland the prefix
could be 4860 (in reality there are more cell phones’ prefixes in Poland, this example is a
simplification). A new entry in the “dialing plan” can have the following form
“telephone number” – 4860
“priority” – 0
“external gateway” – gateway 3
“signaling proxy”
“tech prefix” – 0
“dial as” – 60
“media wait for connect” – inactive
“IE Display from client” – active
After completing this entry all the calls to the number e.g. 48600316058 will be directed
to the Gateway 3 in “signaling proxy” mode. The number will be changed and have the
form of 0600316058. In this form the number will be sent to switch (“dial as” = 60
results in cutting “48” in “4860” prefix, “tech prefix” = 0 results in adding “0” in front of
the number)
If, for any reasons, connection can not be completed VoipSwitch will try to send the
connection to the gateway entered in the “dialing plan” with the higher level prefix i.e.
less detailed, in this case “48”, and corresponding gateway 1.

In the “telephone number” field, end users’ numbers can also be entered e.g.
48600316058, with the “internal gatekeeper” option active and the client’s name selected.
It will result in sending all the traffic directed at this number to the client’s device,
provided the client is currently logged in as a gatekeeper. In case this connection fails it
will be redirected to gateway 3.
It is also possible to add characters or numbers to the “tech prefix” field, which will be
placed at the end of the telephone number in the calls coming out from VoipSwitch e.g.
entering [n] # will place “#” at the end on the number coming from the client. For
instance “00[n]# “ can be entered in the “tech prefix” field in “dialing plan” for the prefix
48 (“telephone number” = 48). As a result, if a client sends the connection to the number
48600316058, VoipSwitch will change the number to 0048600316058# and send it to the
final gateway.

Types of proxy modes

System VoipSwitch allows sending connections in various proxy modes, which are set
for the particular prefix in the “dialing plan”. Thanks to this, different types of
connections can be associated with different types of clients. For instance, calls from the
callback and pc2phone clients require “proxy all” mode and will be sent with the
appropriate prefix e.g. 48, while VOIP gateway clients can connect in “signaling proxy”
mode, sending unique prefix in front of the telephone number e.g. 33348…(unique prefix
can be added to the calls, coming from a particular client also from the clients’
description level. Thus, it is not necessary for the clients to add a unique prefix by
themselves (more information on this issue further).
1. “proxy all, connect independently” – in this mode a client establishes connection with
switch and negotiates with it. Afterwards, switch connects with the target gateway
and also negotiates characteristics of the connection. After establishing all of the
characteristics of the connection this information along with the voice packages goes
through switch – this process is described as a full proxy.
2. “proxy all, forward call signaling channel and h245 channel” This is also full proxy
option. The difference between this option and the previous one is that the call setup
from the client is sent to the target gateway. Also, information coming from a client
through h245 canal is directed by switch to the gateway and then to a client.
3. “proxy call signaling channel and h245, no media proxy” – in this mode only
signaling information and h245 canal go through switch, media packa ges are sent
directly between endpoints.

Endpoint selection for the give prefix

Calls coming to the particular number can be directed to one of the following:

• External VOIP gateway – “gateway” option needs to be marked and

one of the gateways chosen from the gateways list defined in the
“gateway” menu. Voipswitch will send the call to the target gateway.
• External gatekeeper – “external gatekeeper” option needs to be marked
and one of the previously defined gatekeepers chosen. VoipSwich, if
currently registered to the particular gatekeeper, will send calls there.
• Internal gatekeeper – after choosing this option the list with the clients
that register to Voipswitch as to gatekeeper will be displayed.
Choosing the client from the list will result in Voipswitch sending calls
for the particular number to the device registered in Voipswitch. For
instance, if a client possesses Cisco ATA and is registered in
Voipswitch, the telephone number can be assigned to the client and
calls can be transferred there.
Using this method it is possible to send connections to gateways/IP
phones that use dynamic IP. Call will be transferred only if a particular
device is currently registered, if not dialing plan can be set up so the
calls are redirected e.g. to voicemail or different telephone number.

Managing the tariffs

VoipSwich enables adding unlimited number of tariffs, which can be assigned to clients.
Different tariff can be assigned to different clients. In order to create a new tariff menu
“tariffs” has to be entered, name for a new tariff established, minimal duration and
resolution parameters have to be defined and “add” button clicked to approve selection.
The new tariff will appear in the tree chart. Properties of each tariff can also be changed
from this level by choosing a particular tariff, changing its resolution and/or minimal
duration and approving it by clicking on “change” button.
“Minimal duration” – denotes the minimal time that the caller will be charged for. For
instance if it is set on 30 seconds then the call that lasted only five second will still be
charged for 30 seconds.
“Resolution” – also described as billing step, denotes the minimal step (measured in
seconds) according to which connection cost is calculated. For instance if the
“resolution” is set to 6 seconds and the call lasts 55 seconds, cost will be calculated as
for 60 seconds (10 full steps, 6 seconds each makes up 60 seconds). Time that a client is
actually charged for is calculated as follows; Call time is divided by the number set as a
“resolution” rounded up to whole number and multiply by the number of seconds in a
resolution unit (in this case 55/6 = 9.1 (10); 10*6 = 60) if we want to charge the clients
for the actual calling time then “resolution” has to be set on “1”.

Picture 5 Tariffs menu

Created tariff has to be completed with setting up rates for the particular prefixes.
Description, price per minute and prefix have to be indicated along with the days of the
week and hours within a day that the particular tariff refers to. For instance:
Description: Poland
Voice rate: 0.02
Prefix: 48
From day: Sunday
To day: Saturday
From hour: 00:00
To hour: 00:00

Description: Poland cell

Voice rate:0.20
Prefix: 4860
From day: Sunday
To day: Saturday
From hour: 00:00
To hour: 24:00

These two entries will result in charging the calls directed to the numbers starting with
4860 according to the tariff set for “Poland cell” i.e. 0.2. All the other calls starting with
“48” e.g. 4822 will be charged according to 0.02 rate. Both rates are valid throughout the
whole week, 24 hours a day. Using the “time span” options, different rate can be set for
different days or parts of a day e.g. different rates can apply to weekdays and different to
weekends. Also, different rates can be set for a pick time hours during the day and off-
pick time hours. To diversify rates within a week and a day “time span” for a particular
prefix has to be set accordingly.
Important: client can connect only with prefixes that are defined previously for a
particular tariff. If, for example, the client is to be limited to make calls only to prefixes
representing e.g. Poland, the tariff that is assigned to the client should only include
prefixes representing Polish telephone network, i.e. numbers starting with the country
code 48. Calls from this client directed to different prefixes will be rejected.
Also “disable this prefix” option can be used to limit client’s access to the particular
prefix. For example prefix 48700 (paid services).
Picture 6 Managing a tariff.

Importing tariff

It is possible to import a particular tariff together with applicable rates from the tariff
level. To accomplish that, suitable text file has to be prepared and “import” button
clicked with choosing destination on the hard drive.
The text file should have the following format.

355;Albania;0.203;0;6;0;2400
213;Algeria;0.194;0;6;0;2400
2131;Algeria cellular;0.189;0;6;0;2400
684;American samoa;0.119;0;6;0;2400
376;Andorra;0.069;0;6;0;2400
3763;Andorra cellular;0.163;0;6;0;2400
244;Angola;0.318;0;6;0;2400
1264;Anguilla;0.210;0;6;0;2400
1268;Antigua/Barbuda;0.264;0;6;0;2400
54;Argentina;0.066;0;6;0;2400
5411;Argentina b aires;0.040;0;6;0;2400
549;Argentina cellular;0.107;0;6;0;2400
The columns denote (starting from the left)
Prefix, description, voice rate per minutes, from day, to day, from hour, to hour

Note: entries in this file should start in the first row, without entering columns’ headings.
“description” should not exceed 25 characters.
Columns ha ve to be separated with the semicolon (;). This type of file can be prepared in
Excel and save in a CSV format.
If there is an error in the files, VPCConfig will stop the import process and identify the
error. Records from the imported file will be added to existing tariff. If existing rates are
to be changed they have to be removed first (“remove all” button) and only then file can
be imported

Clients’ accounts management

Adding and managing clients’ accounts is performed from the level appr opriate for the
particular type of service.
The following groups of clients are identified in VPSConfig:
• GW clients
• GK/Registrar clients
• Pc2phone clients
• Callback clients
• PIN clients

The first group, GW clients , includes users’ accounts that send connections directly to
the h323 port or VoipSwitch’s SIP port. Clients can comprise of h323/SIP gateways,
Ipphones and other h323 terminals or SIP that connect to VoipSwitch in peer-to-peer
mode. In this type of connection VoipSwitch functions as h323 gateway or SIP proxy
depending on the protocol type that is used.
For this type of clients authorization can be conducted based on the IP address of the
device that is sending the call, h323, or, in case of SIP, a pair of login and password. In
authorization based on IP, several addresses can be assigned to one client. For instance
when a client has several h323 gateways from which traffic is sent to VoipSwitch.
It is particularly useful in wholesale selling of telephone traffic.
Authorization based on h323ID is useful when the client’s device operates on the
dynamic IP address.
Picture 7 GW clients menu

"GK/Registrar clients" group is also designed for the end users, calling from the
various IP addresses. The difference between this group and GW clients is in the way
connection is carried out. In case of GK/Registrar clients, device (e.g. SOHO gateway, or
Ipphone) has to register to the gatekeeper first, and only after the successful login can
send calls. In this mode VoipSwitch functions as an h323 gatekeeper or SIP registrar,
depending on the protocol.
Authorizing h323 device that registers to VoipSwitch’s gatekeeper is based on the alias
sent. In case of SIP, authorization is based on verifying login and password.
Authorization based on h323ID or alias (in GK mode) requires entering a login/password
string in appropriate field in device’s settings. The string has to be divided by a
separating character that is defined in “h323 settings” in VPSConfig – by default it is
“@” character. Exemplary login/password string could have the following form:
chris@test, where “chris” denotes login, “test” denotes password and “@” is a separating
character.

Following menus are designed for managing pc2phone and callback accounts. In order to
allow pc2phone users to utilize callback services using the same account, the option
“Allow pc2phone users log as callback users” has to be active. This option is located in
“VoipSwitch settings” menu in VPSConfig.
Shared accounts’ properties for each type of client

Regardless of the type of service, the following parameters have to be established for new
accounts.
Login
Password
Account type – prepaid or credit
Current amount – the amount assigned to a client. In case of “prepaid” type of account,
the amount has to be set in advance e.g. 100 – this amount will be reduced by the cost of
each call made by the user. If the current amount is less than the billing step then
VoipSwitch will automatically end the connection.
In case of “credit” type of account, the current amount shows “0” at the beginning. The
cost of each call is added to the current “credit” amount.
Active – user can connect only if this field is marked
Tariff – out of previously created list a tariff has to be chosen that the user will be
charged with.
Dialing plan prefix – this option enables altering the phone number that the client wants
to connect with. For instance entering 777 will result in adding 777 in front of the
telephone number. In consequence, VoipSwitch will look for the entry in “dialing plan”
corresponding with the previously entered (changed) number. This options enables
differentiating “dialing plan” so the calls made by different users to the same number can
be sent to different target gateways and/ or with different connection attributes. To
illustrate this consider the following: routing for pc2phone and callback clients is set and
their calls are sent to carrier 1 in “proxy all” mode. If, at the same time we want one of
the clients who has VOIP gateway to send the traffic in “signaling proxy” mode, “dialing
plan prefix” can be set to e.g. “*111”. With the above-described setting pc2phone and
callback users, calling Poland, will dial 48*******. Those calls will be directed
according to the entry made for prefix 48. As for the client who had “*111” added, the
new entry has to be made for *111 prefix. When this client will try to connect with the
number 48600316058, VoipSwitch will automatically add *111 in front of the number
and will search for the corresponding entry in “dialing plan”. In this way new
connection’s attributes e.g. “signaling proxy” can be set for *111 entry and it differs from
the settings set for all the other clients.
Additionally, using “dialing plan prefix” the initial digits in a telephone number can be
dropped. For instance if clients sends calls in “00 + country code” format, and the whole
dialing plan is set without “00” at the beginning, then “00” can be dropper from every
call coming from this client. In order to accomplish that “00->” has to be entered in
“dialing plan prefix”. If, for example, “00” needs to be switched to “11” the following
entry has to be made:
“00->11”
Changing the number that the connection is to be established with (called number) does
not change the rate the connection is charged with. If the client calls 48600316058 and
this number is changed to 1148600316058, VoipSwitch will still bill this client according
to the incoming call i.e. for the prefix from Poland – 48, not 1148
Changes made in “dialing plan” field only constitute principles used by VoipSwitch
while exploring “dialing plan”.
Tariff prefix – similarly to “dialing plan prefix” this field enables altering telephone
number. The changes only affect the way rate is selected within a tariff. For exa mple if
the rate is created for the 48 prefix and the client sends the number in 0048 format, it is
possible to drop “00” by entering “00->” in “tariff prefix” field. In the same way if the
client sends the number in 600316058 format, without the country code, then 48 can be
added and VoipSwitch will charge this call according to the rate for 48 prefix.

If we want the changes in called number to affect both selecting the rate within a tariff
and selecting the appropriate routing in “dialing plan” then exactly the same entries have
to be made in “dialing plan prefix” and “tariff prefix”

Note: for callback clients there are separate “dialing plan prefix” and “tariff prefix”
fields for connections with both source and destination.

Setting up voice codecs

In the “GW clients’ and “GK/registrar clients” it is necessary to enter the parameters that
would define how the calls coming from a particular client will be coded. There are two
options: g723.1 codec group and g729 codec group. These settings are necessary only for
“proxy all” mode. If the other mode is chosen it is also possible to send calls with
different codecs (important here is that the client’s device and target gateway use the
same codecs). In “proxy all” mode, client’s device connects with VoipSwitch and then
Voipswitch connects with the target gateway preventing negotiating codecs between the
endpoints. Lack of possibility of negotiating codecs in “proxy all” mode reflects the
improvement in the speed of operations. For example, by setting up g732. 1 codec for the
particular gateway, VoipSwitch recognizes that the final connection should also be
carried out using this codec. That enables VoipSwitch to search the appropriate gateway
in the routing table and carry out the connection. If both g732.1 and g729 are supported
by client’s device then one codec can be set as a primary and the second as a secondary.
As a result VoipSwitch will try to connect with the gateway working with the primary
codec first, and if this attempt fails, to connect with the ga teway working with the
secondary codec. In case a client has two codecs set up, his/her device has to have
capabilities of negotiating codecs.

It is not necessary to set up codecs for “pc2phone clients” as the software provided with
VoipSwitch works only with g723 codec and there is no option of changing it.
Automatic Users Generation

Note: refers only to pc2phone, callback and PIN (2 stage dialing) clients

This menu enables creating and managing “lots” that are groups of automatically created
clients’ accounts.
The window is divided into two parts: upper for managing already existing “lots” and
lower to create new clients’ accounts. “lot” can be chosen from the list in the upper part
of the window and either activated (all the accounts that belong to the lot will be
activated), deactivated, deleted or exported to a file (the text file will be created with the
login/password pairs and serial numbers)
In order to create a new lot its name has to be created and parameters chosen, based on
which login/passwords pairs will be generated (login/passwords can contain letters and
numbers or letters only - capital and small letters or small letters only). Furthermore types
of accounts have to be chosen - pc2phone or callback – and rest of the fields have to be
filled out, similarly to creating a new account in “pc2phone” or “callback” menus
The operation is approved by clicking on “Create a new lot” button, after which new
accounts will be generated.
Picture 8 Automatic users generation

Menu Calls and Failed Calls

Menu “Calls” displays the list of all the calls made in a given period. For each call there
is detailed information on the duration of call, cost, name of the client, route the
connection utilized and many other useful information. By using filters only the desired
records can be displayed e.g. in order to generate CDR for a particular client. “Export”
button enables saving records to csv, xls, html or xml files.
Picture 9 Exporting calls table

Below the list there is information on total cost of all calls, average cost of a call, total
time of calls and average time of a call.
Picture 10 Calls menu

“Failed calls” menu is organized similarly with the exception that it presents different set
of information. Connections that have failed, for various reasons, are displayed here.
Description of errors and release reasons helps identifying the potential problems
Picture 11 Failed calls menu

Statistics

In the “statistics” menu all the connections are displayed both successful and failed in a
given period of time along with ASR, ACD and PDD indexes. It is possible to limit the
statistics to the selected, desired information e.g. calls sent only to the particular gateway,
particular prefix or statistics referring only to one client.
Picture 12 Statistics
Picture 13 Statistics caclulated for prefix 44