Softswitch FAQ


Too low ASR


1. Check disconnect reasons for that direction
2. Check if gateway audio is ok


SIP user cannot be called


1. Check disconnect reasons in cdr record for that called
3. Check if username exists
4. Check if usergroup matches the caller usergroup
5. Check user firewall settings


How to add an SMS provider?


The server can connect to SMS providers (such as clickatel) and send SMS message via HTTP GET request.
The URL parameters must be obtained from the SMS provider.
Set the URL in the global configuration under the “smsurl” key.
You can use the following keywords in the URL:
[fromid], [fromnum],[tonum],[message],[smsid]
After all SMS message a CDR record will be created with it’s type set to “SMS”.
For unified pricing you can use the “smsprice” and the “smstime”. (in this case all sms messages will be billed with the same price, regardless of the destination)
Othervise the pricing can be set on the “Price Setup” form the same way as for voice calls. The only difference is that all SMS messages appear with 60 sec duration.
Unicode messages are supported.


How calls are processed


1. The SETUP or INVITE signal arrives from the traffic sender
2. If the caller is not allowed by the firewall, the call will be silently dropped
3. If the caller is blocked (e.g. DOS attack protection), then call will be silently dropped
4. Caller authorization (by source IP address, username/password, techprefix, etc)
5. Check the call parameters. If doesn’t fit into the predefined limits, the call will be dropped (example: too long called number)
6. Rewriting the called number if any Prefix Rule Match
7. Normalizing the called number (validating call prefix)
8. Searching for the best routing pattern
9. Searching for best route direction (available channels, priority order, round-robin, LCR, BRS, failovers, rerouting. etc)
10. Calculating the maximum speech length based on caller credit
11. Checking class 5 features and other endpoint settings (media routing, early-start, etc)
12. Initiating protocol conversion if needed
13. Routing the call to destination
14. Checking for call status, dropping if time exceed and other call monitoring tasks
15. Collecting CDR records at the end of the call
16. Calculating the prices of the call (realtime billing)


NATs and firewalls


There are several ways UDP might be handled by a specific NAT or firewall implementations, these are categorized into:

Full Cone NAT

A full cone NAT is a solution, where all requests from the same internal IP address and port are mapped to the same external IP address and port. Furthermore, any external host can send a packet to the internal host, by sending a packet to the mapped external address.

Restricted Cone:

A restricted cone NAT is a solution, where all requests from the same internal IP address and port are mapped to the same external IP address and port. Unlike a full cone NAT, an external host (with IP address X) can send a packet to the internal host only if the internal host had previously sent a packet to IP address X.

Port Restricted Cone

A port restricted cone NAT is like a restricted cone NAT, but the restriction includes port numbers.

Specifically, an external host can send a packet, with source IP address X and source port P, to the internal host only if the internal host had previously sent a packet to IP address X and port P.

Symmetric Nat

A symmetric NAT is a solution, where all requests from the same internal IP address and port, to a specific destination IP address and port, are mapped to the same external IP address and port. If the same host sends a packet with the same source address and port, but to a different destination, a different mapping is used. Furthermore, only the external host that receives a packet can send a UDP packet back to the internal host.


What ports are used in the system?


Standard SIP signaling port: 5060 (TCP and UDP)
Default H323 signaling port: 1720 (TCP)
H323 signaling port used by Mizu gateways: 1721 (TCP)
Rdesktop port: 8836 TCP
SQL Server port: 1433 or 2223 TCP
“Voice Here” port: 44444 UDP
Mizu server admin port: 9885 TCP
Mizu server comm. port: 9886 TCP
Mizu server log  port: 9889 TCP
Virtual SIM port: 9886 UDP
H323 additional port: configurable dynamic TCP
Media ports: configurable dynamic UDP
WebServer: 80 TCP
FTP: 21,22 TCP
Enduser/Callshop/Reseller WebPortal: 8080 or 8082 TCP


Sometimes the voice channel is breaking down. How can I improve the voice quality?


Set up to active silencedetection (silencedetection=1)
Increase the jitter buffer (minjitter, maxjitter)
Use a low bandwidth codec (onlyg72x=1)


 How to check the call quality on a specific channel?


1. In the “Set Directions” box set the preffered simid. Then go to the “Statistics” form and check the ASR/ACD values.
2. Start some tescalls (right click on the preferred channel and then hit the “Test Call” menu)
3. Listen to conversation. (“Voice Here” form)


How can I make test calls?


1. simply right click on a channel (“Simcards” form) and select the “Test call” option
2. or use one of the voip clients from the “Tools” menu


Mizu Server security


OS security:
Follow the instructions below to secure your VoIP environment:
-do not install any third party software on your VoIP server
-enable the embedded windows firewall. It’s speed and application level packet filtering is perfect for VoIP. Enable only the needed applications (voipserver.exe, mssql.exe, vfpt.exe)
-always change the default MSSQL port from 1433 (this is not a security issue, but when the MS SQL runs on the default ports, you can experience lots of login attempts). 
-disable all unneded network services (IIS, FTP, etc)

The mizu server has the following builtin automatics attack prevention mechanism:
Address level attack preventions:
-DOS attack prevention: when there are too many messages received from an IP address, the address will be blakckisted automatically. Controlled by MAXSUBSMSGCOUNT and MAXSUBSMSGPERIOD global configuration settings.
-when there are too many “wrong” or meaningless messages from an IP, the address will be blakckisted automatically. Is controlled by MAXWRONGMSGALLOWED global configuration setting.
Session level security:
The server will close the session on the following  circumstances:
-absolute timeout, call timeout, media timeout, ring timeout, call init timeout, timeout  on session timers
-too many incoming messages
- too many authentification failures
- too quick (abnormal) message receptions

Blocked devices and users can be reenabled anytime by issuing the “delbanned,ip” or “delbanned,all” command on the Console port.

Device IP/user caching:
For speed considerations, the mizu server can cache device login information for a time so it will not ask again for authorization for every REGISTER or INVITE request. This means that it can happen that you change the user credentials and the user is still able to login with it’s old username/password or inverse: the user enters the correct credential but will be still blocked for a time. This can be controlled by the “cacheregistrations” global config variable.


What are the minimal global settings that must be correct on servers?


On the “Configuration” form select “Basic” settings and check at least the following values:
LocalIP, LocalInternalIP, LocalDomain, currency, Routing, emergencydir, creditunit


No voice (caller and called cannot hear each-other)


1. Check routertp settings for the caller and the called
2. Check called firewall and nat settings


How to setup a new virtual server instance?


1. create a new directory named “virtserverX”
2. copy all files from an old virtserver directory (except the logfiles)
3. rename virtserverOLD.exe to virtserverNEW.exe
4. register the new virtserverX.exe as a system service
5. rewrite the start.bat and stop.bat
6. allow the new service (and other executables if any) in the OS firewall
7. create a new ftp directory for the recorded voices files under the “voice” directory and set the correct rights
8. set the database connection settings in the inifile
9. clone an old database instance
10. clean old database tables (MManage ->Advanced->Clean Database Tables)
11. setup default tables (MManage ->Advanced->Setup Default Virtserver Database)
12. upgrade to the newest version (MManage ->Advanced->CC SW Version Migration)
13. setup configuration (MManage ->Settings-> Quick Setup)
14. check port collisions (rewrite portnumbers)
15. configure routing in the mainserver (in and out). Add the virtserver user as traffic sender and sipproxy
16. allocate numbers in the mainserver for the new virtserver instance (callback numbers)
17. start the new virtual server
18. make a testcall
19. check the logfile for errors. test MAgent. check MManage


 How to add a new sip enduser?


In the “Users and Devices” form select Endusers. Load the list and then hit the “New” button. Then you have the option to clone an already existing traffic sender. Set up the authorization correctly! Check the credit and prepaid/postpaid option!


Working with resellers


-set the “resellerbilling” global config option to true
-unlimited reseller child/parent relationship can be created (limited by the “maxresellers”  global config options)
-this relationship can be analyzed using the “Ownerships” form,
-make sure that you have a public reseller price listing
-reseller will be able to create their own prices on the website
-reseller can create a “base tariff” and other tariffs assigned to individual users
-individual reseller prices are stored in tb_billsources with their “resellerid”
-if reseller has not tariffs, than the billing will be done after usual enduserprices
-top reseller id is stored in tb_cdrs.resellerid and the “othercost” field will contain the payment from the reseller (loaded from public reseller price)
-individual reseller cdr records are stored in the tb_cdrresellers
-check reseller statistics on the “Advanced Statistics” form


VoIP Encryption


Optionally a separate udp port can be set on the server to handle encrypted sessions (alternatelocalport)

Encrypted sessions are always anwered encrypted by the server.
Use the tb_users.encrypt field to control encryption on user level:
  2=dynamic no
  4=dynamic yes
  6=force always
To initiate encrypted sessions, set the “encrypt” field for the user to 5.
To deciede automatically, then set to 2.
To encrypt all communication on the alternatelocalport, set “alternatelocalportencrypt” global config to 2.

There are two types of builtin encryption:
-weak and quick encryption mode (when usequickencryption is true -default)
-storng but slower encryption mode (when usequickencryption is false)

Encryption can be enabled/disabled with the “useencryption” global configuration value:
1=only when rec encrypted (default)

Also, there is a possibility to define a list of ip addresses with the “encryptedpeerlist”. All communications with these peers will be done encrypted.

Standard SSL/TLS signaling encryptions are negociated runtime as well as SRTP media encryption.
The recommended encryption type between mizu devices is the embedded fast or bowlish+compression. These types of encryption require much less CPU power and are done with no additional network overhead.


Running Mizu Server in proxy mode


(the server doesn’t handle registration and authentication, just forward it)
Fwdregistrations=2 // //0=no,1=only from alternate port, 2=always
fwdregistrations_domain=registrar sip domain
fwdregistrations_ip=registrar ip or FQDNS
fwdregistrations_port=registrar port
autocreatereguser=1 //0=no,1=when fwd authenticated ok register, 2=always (when we receive the register)
Forwardauthentifications=1 //will forward invite (regarding routing setup)
Maxreroute=1 ??

Setup routing to point to the upper server

Optionaly you may increase these tresholds:
Checkmaxlines = 0 ?
Checkmaxlinetb=0 ?
Maxsessionspeechlen= 1000UL * 60UL * 60UL * 6UL

Running the server as a VPN access point
Fwdregistrations=1 //only from encrypted clients (bug: set to 2)
Alternatelocalportencrypt=3  //0=default,1=never,2=auto,3=always
Set the “encrypt” field for the users to 5.

When all the new automatically created users are allowed to use only encrypted comunication, set the autonewusersencrypt to 5.


Typical Cisco Config


! dial-peer voice 3630 pots incoming called-number 0040T direct-inward-dial port 2:D ! dial-peer voice 3631 voip destination-pattern 0036 voice-class codec 1 voice-class h323 1 session target ipv4:


Basic callcenter tasks


1. Setup your server as for a normal sofswitch (routes)
2. Create campaigns
3. Add callcenter operators
4. Assign operators to campaigns
5. Add or import clients
6. Assign clients to campaigns
7. Add presentation locations
8. Setup global callcenter configurations
9. Operators now are ready to start there MAgent application
10. Check statistics
11. Print invitations
12. Use checklist when you are on presentations


CLI settings / A and B numbers / Dial plan


There are multiple ways to apply your rules.

The server will “normalize” the numbers automatically based on rules from the global configuration (search for normalize). For exampe it can remove strange digits, IEC codes like 00 or +,  enforce length, etc.
Also there are built-in rules for several countries, so users can dial without CC/IEC/NEC.

For outgoing calls if you need to use a tech prefix, you just have to enter the coresponding digits as tech prefix for the sip proxy user.

The server can authorize and/or route the traffic after the incoming techprefix.
Sip users can have techprefixes too. this is usually common for reseller company users.
If no techprefix is specified, then it will be loaded from tb_pxrules if any.
Sim owners and vpc users can have a list of prefixes separated by comma.
If no techprefix is specified, 111 will be inserted for incoming called numbers.
If the techprefix is „-1”, then the original techprefix will be forwarded.
If the techprefix is „-2”, then the original techprefix will be inserted in cdr record (but not forwarded).
If the techprefix is empty, then only the normalized callednumber will be forwarded.
The following techprefixes are reserved for the server: 111,222,999.
Only 3 digit techprefix is allowed. If your traffic sender needs another techprefix length, you must rewrite the incoming number in the “Prefix Rules” form.
            Example: protcoll: sip, Type: ip, value: your traffic sender ip, rewritefrom: oldtechprefix, rewriteto: newtechprefix.

If you need more complex rules then you can use the prefix rules form in the mmanage application.Here you can add/remove any prefix for A and B numbers before and after the routing.

You can rewrite prefixes before they arrive to the routing by entering your preferences here.
The Mizu routing engine will accept only 3 digit length techprefixes or no thechprefix, so you must convert them here if your traffic sender will send the traffic with techprefix that are not three digit length.

For example you can set up a rule which defines that every incoming number from ip on H323 if begins with 1234 must be rewritten to begin with 56.  Number 123499999 will be rewritten to 5699999.
If the “RewriteFrom” is emtly, then the “RewriteTo” fill be insterted before the number

To rewrite prefixes on router number normalization, you have to set the following global config values:
prefixrewritestr: the original prefix
prefixrewritefrom: keep from
prefixrewriteto: insterted string
for example to handle the hungarian roaming prefix: 08 + SK + BK + NSN +SN you have to set the following values:   
prefixrewritestr: 08X…
prefixrewritefrom: 9
prefixrewriteto: 36

If you need more flexibility, you can edit the v_check_pxrules stored procedure manually using any SQL expression.

For CLI control you can set the “cli” field for any user:
CLI:   CLIR and CLIP settings
    0: forward always   (forward asserted as normal number always!). Will not hide, even if caller was set so.
    1: normal handling (forward asserted as normal number) -default
    2: forward as asserted identity always (identityrewrite asserted)
    3: forward as asserted identity only to trusted domains (identityrewrite asserted)
    4: normal hide (no idenityrewrite forwarding)
    5: force hide (no asserted identity too!). Always hidden.

To completely rewrite the A number you can use the “rewriteanumber” field.
To cut some prefixes from the A number use the “cutanumber” field.
To allow some A numbers and rewrite other numbers you can use the identityrewrite and identityforward fields.
This can be useful when not all your user have real PSTN numbers.

Addtechprefix: we insert this number before the callednumber if the caller doesn’t send its calls with tech prefix.
identityforward: we can toward these kinds of usernames and the other we rewrite to „identityrewrite”.
identityrewrite: if the caller username don’t match the identityforward prefix, then we rewrite it.

For more control over the A number you can use the prefix rules form or manually edit the v_check_pxrules stored procedure.

Prefixed can be also used for authentication, routing and billing.

Q21. What is Mizu Softswitch?
Answer: Mizutech Softswitch is a comprehensive integrated, high-performance, scalable and robust SIP platform for delivering Internet Telephony (VoIP) services to all kinds of customers - from residential home users with one or two lines, to business and wholesale clients with many thousand channels.


How to add endusers (basic settings)


1. Go to MManage -> Users and device form, and select enduser type
1. Select an already existing user wich has the same caracteristics as the required new endusers
2. Hit “New User” and than accept the the copy from existing option (cloning)
3. Check at least the following fields: username, password, parent id, authorizaton type (usually username/password), prepaid/postpaid, billed user
4. Check other settings
5. Save


How to disable CLI for all outgoing calls


For the sip proxy users(s) in the rewriteanumber put “anonymous”. Make sure you don’t have identity forward and identity rewrite (both of them must be emty)


SIP caller cannot call


1. Check disconnect reasons in cdr record for that caller
2. Check username/password
3. Check credit (if prepaid user)
4. Check caller techrefix, and the routing settings for that techprefix


Fax settings


faxhost: email-fax gateway domain name or ip
faxuser: smtp username
faxsubject: email subject
faxfromaddr: from field
faxfromname: from name field
    0 = leave original number
    2=normalize, no iec
    3=normalize, with iec

faxsuffix: append after faxnumber (usually @host.domain)


How to set up the automatic credit recharge?


First you have to set up the “Message Rules”.
The packet must be set to prepaid. Proper Credit Request/Charge command must be defined. See at 4.6.1. SIM Packets
SIMcard “Credit and Recharge” setting must be set accordingly.

Message Rules types:
    -0: msgbgn need to be replaced with msgend before further processing
    -1: if msgbgn was found in the sms than it means a successful recharge
    -2: credit value between msgbgn and msgend
    -4: if msgbgn was found in the sms than it means a failed recharge


How to play user credit in IVR


Create a “Play number” action.
Enter [credit] for the file name.


ATA problems


If there are no message received on our server (search for ip or username in the last server log):
-ATA config error (reset to factory default and retype the account info)
-router/firewall problem: try the same device behind another router or from a public IP
-the IP is rejected by our server: you can check the blocked device by opening the “Server Console” form, connect and type “bannedlist”. You can clear the list with the “delbanned,all” command
-try the same ATA with any another VoIP server

If messages are received by the server
-then most probably we can find out the problem from the server logs


How to add a new traffic sender?


In the “Users and Devices” form select Traffic Sender. Load the list and then hit the “New” button. Then you have the option to clone an already existing traffic sender. Set up the authorization correctly!


How to restart  the server box?


-MManage->Administration->Server Console->Connect  and send the „pcrst” command
-If you cannot  connect with MManage, you can find a small program in the vclients directory named „serverrst” (usually at C:/Program Files/MizuVoIPServer/ serverrst.exe
-If these does’nt work, then the server has a serious problem. Follow the failovering plan and call the administrator


Multihomed setup


UDP ports must be binded to different ip interface (at least the default sip port 5060), or set to different values.
TCP ports must be setup differently
On the gsm gateway set the serverip to the binded ip and the  cmdport to the gsmclientport
Change the pipename setting in the gatekeeper.ini


Working with callshops


-callshop owners are endusers
-cabins are sub-endusers where the parented and the billed user points to the callshop owner
-callshop owners (endusers) can add/modify their sub-endusers (cabins) from the web

Q33. What kinds of business scenarios are supported?
Answer: The Mizutech Softswitch is suitable for the following business scenarios:
-wholesale traffic
-VoIP call termination
-SIP SBC or proxy server
-Vonage-like retail operations
-calling cards and callback services
-hosted call shops
-multi-tier traffic reselling

Q34. Is it for me?
Answer: If you are just starting a new Internet Telephony business or willing to upgrade your existing VoIP platform to a professionally developed and supported solution without being charged arm and leg, then this is a right product for you.


Short number and internal billing


You can set the short number billing mode by the “shortnumbill_type” config value. When set to 0, all short numbers are billed with the price set in “internal_providercost” and “internal_endusercost”. When set to 1 (default),, then the  “999” prefix is inserted before the called numbers, and you have to create billing entries for these numbers.
Numbers are treated as “short” if its length doesn’t exceed the value set by the “billshortnumlength” global config option.
Calls between endusers are billed with the “internal_providercost” and “internal_endusercost” values (defaults to 0).


How to change the database username/password


The auto installer will use the “sa” username with “srEgtknj34f” password by default.

Install the SQL Server Management Studio if not already installed.
Login with the old credentials.
Open “Security” -> “Logins”. From here you can easily add/remove any users or change passwords.

Before changing the default user settings, stop the mserver service (By launching the stop.bat file or from Services).
Change the database settings in the mizuserver.ini configuration file.
Restart the voip service.


How is VAT handled?


You should try to use prices without VAT included all ower in the system (for pricelist and for simcards)
VAT included pricelists can be easily converted to net values by checking the “Convert to NET value” checkbox in the “Price List”. You should enter the VAT percent in the “VAT Value” editbox for proper calculations.

For simcards you can setup the VAT value in the Packet options (“VAT” editbox). If you set the “convertsimcredittonovat” global configuration options to true, than sim credits will be automatically converted to net values. For examlpe after an automatic credit request, the credit value in the received messages (SMS) will be automatically conveted to net values.

You should set up the appropiate VAT values for users too, wich will be taken in consideration during the billing process.


Too slow MManage


1. Check your internet connection
2. Check server processor load. If too high, then check server logs, and if necessary, restart the server
3. If the problem persists, call the administrator


How to add plugins in MManage


Create your custom exe.
In the MizuManage.ini under the “plugins” section add your plugin entry:
X is a number starting from 0 to MAXPLUGINCOUNT

The MManage will pass the database access parameters in the command line.
Any other parameter must be readed from the inifile.

Command Line: dbserverip,dbport dbname username password connectionstring
*Please note that appserverip may differ from dbserverip!
Direction parameters will be located under the “parameters” section. Read them as soon as possibile.
The following keys are defined:
    fromdate, todate
    source, destination
    fromtype,fromid, frompacket, fromsimid, fromgroup, fromcamp
    totype,  toid,   topacket,   tosimid, togroup
    fromname, frompattern
    toname,   topattern
    fromenabled, toenabled
    fromidlist;    toidlist
    fromsimidstr;  tosimidstr
    fromip, toip
    fromsimgrouplist, fromusergrouplist
    tosimgrouplist, tousergrouplist
    reloadgroups reloadgroups


How to restart the server service


1. Login to server remode desktop. Use start/stop commands from the start menu or open the windows services, right click on mserver and choose stop, then start
2. MManage->Administration->Server Console->Connect  and send the „servicerst” command


Where can I check the logs and traces?


1. “Logs” form
2. “Server Monitor” form
3. Set up your trace level in the “Configurations” form (filer after the “log” expression)


How to redirect or forward sessions to other domains?


Routing to other domain can be restricted by the “fwdtootherdomains” global config setting. This means SIP calls with other domains in the request uri (not listed in the “domainnames” values, and no IP address match)
 The following values are defined:
0=don’t forward calls to other domains
1=check if our numbers first (local user)
2=don’t forward mobile numbers
3=forward all required sessions
4=forward all required with “Moved Temporarily”
5=unconditional move (will move all required traffic without checking the caller)

You can forward calls to other domains before any routing configuration will be checked. (with 302 Moved Temporarily)
The following config values will be applied:
Forwardpx:  comma separated prefixes to be forwarded. ‘*’ means all traffic (can be used for failowering and load balancing). Set to emty to disable forwarding.
Forwardto: IP address or domain name where these prefixes will be forwarded.

This feature is usually not enabled for service providers ( if you provide services for costs), and usually is enabled for home or company users.


MManage cannot connect to the server


1. Ping the server box. If ping is working, then check your username/password
2. Restart the server if you are sure that it is blocked
3. If still is not working, call the administrator immediately


Working with groups using direct SQL


Open “Direcy Query” from MizuManage or MS SQL Management Studio.

--check related tables
select top 1000 * from tb_groups
select top 1000 * from tb_grouptypes
select top 1000 * from tb_groupentries

select * from tb_groups, tb_groupentries, tb_grouptypes
where tb_groupentries.groupid = and tb_groupentries.entrytype = tb_grouptypes.dirid
--create a new group
insert into tb_groups (name) values ('testgroup')

--add 1 user to the new group (by id)
insert into tb_groupentries (groupid,entrytype,entryid) values
(select id from tb_groups where name = 'testgroup'),  --group id instered previously
1, --entry type. 1 means "endusers" (values from tb_grouptypes.dirid)
100207 --id from tb_users

--add all users to this group which username contains "111"
insert into tb_groupentries (groupid,entrytype,entryid)
(select id from tb_groups where name = 'testgroup'),
from tb_users
where username like '%111%'

--select users from this group
select, tb_users.Username,
from tb_groupentries, tb_users
tb_groupentries.groupid = (select id from tb_groups where name = 'testgroup') and = tb_groupentries.entryid

--delete group
delete from tb_groupentries where groupid = (select id from tb_groups where name = 'testgroup')
delete from tb_groups where name = 'testgroup'


Ringtone for IVR forwarded calls


Can be controlled by the playivrfwdringtone global configuration
We have 4 options here:
0=no ringtone from the server (instead we can play any file directly from IVR). In this case if there is no other IVR file playback in progress, the client can hear the ringtone generated by the called endpoint
1=generate ringtone on ring received if there are no other IVR playback in progress (in this case the client will hear ringtone even if the called endpoint is not generated –i.e. it just sends ringing message in signaling). This is the default setting.
2= generate ringtone on ring received even if other file playback is in progress (stopping the old playback)
3=generate fake ringtone immediately after call was sent to routing

Q46. Can I migrate my existing subscribers?
Answer: Yes
For this you just have to export the users from your current softswitch in a CSV file or Excel sheet.
Billing and routing data can be also migrated.


How to reset a failowered gateway/direction


-set tb_users.nopriority to any date-time in the past
-open failoweing and reset the failowered gateways
-check if all gateways have provider pricing


Automatic prepaid credit expirity


You can set prepaid credit by the “Add with elapse” button to elapse automatically.
The following configurations are defined:
Creditunit: How much credit means 1 day
Maxcreditelapsedays: max number of days when the credit will elapse
Accelapsedays: the number of days from creditelapsedt when the account will expire
Tb_users. Creditelapsedt: date-time when the credit will be expired
Tb_users.Accelapsedt: date-time when the account will be expired


How are different currencies handled?


In the global configuration, a global currency can be defined by the “currency” setting. For example ‘EUR’, and there is the possibility to convert other currencies (used for pricelists, simpackets, users) to this “native” currency.  For prices defined in “Price List” form, there is a possibilty to convert all input prices in “native” currency by checking the “Convert to XXX” checkbox. In this manner for example you can import a pricelist in other currency and that will be converted automatically in native currency when calculating CDR prices.
The conversions are done based on the settings in the “Currency Converter” form. You should update the conversion rates here as frequently as possible.
If you wish, you can leave the original value intact, so you can make your billing in other currencies than the native.
For every simpacket you can also define the currency, wich will affect the simcard credit calculation (automatic simcredit requests and recharges for prepaid simcards). Simcredits can be converted in the native currency format if the “convertsimcreditcurrency” configuration option is set to true. So you can have simcards in different countries, but all simcredits will be shown in the native currency.
For endusers and traffic senders you can also define different currency format in the Users and Devices form, Billing tab. The currency format defined here will be taken in consideration by the billing process.


How to reenable blacklisted but good numbers


- In MManage -> direct query, under the misc section check the “reenable blocked but good numbers” section
- delete old number from the helper table (section 0)
- run the query from section 1. this will load blacklisted but good number. The query execution may take 15 minutes
- list found numbers (section 2) and check it agains the blacklist (section 3)
- now you may delete blacklist entryies or set the “sure” level lower. First check the requested blacklist entry agains the query in section 4 (found numbers may be only a subset from the blacklist entry and in this case you may not delete or modify the blacklist. But if the asr and acl values are good for the blacklist entry, you may delete or modify it). Before you delete or modify the blacklist entry, check the comment (why was that number blocked). Number with comment “jukak” or “autdisabled monthly/weekely/daily” should be deleted or changed without problems.


Simple prefix rewrite


To rewrite prefixes on router number normalization, you have to set the following global config values:
prefixrewritestr: the original prefix
prefixrewritefrom: keep from
prefixrewriteto: insterted string
for example to handle the hungarian roaming prefix: 08 + SK + BK + NSN +SN you have to set the following values:   
prefixrewritestr: 08X…
prefixrewritefrom: 9
prefixrewriteto: 36


H323 signaling problems


Check your firewalls.
Check Gateway Configuration: onlyg7x, connectwithmedia, enableh245tuneling, faststart.




ASR: average success ratio (percent of the connected calls)
ACD: average call duration. The same as ACL
(ACD: Automatic Call Distributor)
ACL: average call length. The same as ACD
SIMID: sim identifier. 13-17 digit number stored in the simcard (and written on the simcard)
IMEI: gsm engine identifier (should be globally unique)
ACT: average connect time. The time elapsed from setup until the connect in seconds
PF: profit. (for correct values, requires your billing module to be properly configured)
SUCC: successful call count (same as ASR but not in percent)
CCC: concurrent (simultaneous) call count
CC: callcenter
TCP:  is a connection-oriented internet protocol
UDP: internet core protocol for datagram packets (not reliable)
RTP: media channel protocol
SIP: The Session Initiation Protocol (SIP) is a signaling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session.
H323: H.323 is an ITU (International Telecommunications Union) recommended standard, which provides a foundation for audio, video and data communications on non-guaranteed Quality of Service networks
RAS: used in H323. Used between the endpoint and its Gatekeeper in order to
Allow the Gatekeeper to manage the endpoint (Registration, Admission, and Status)
GK Registration: Endpoint will send an RRQ and expect to receive either an RCF or RRJ
H225: Call Signaling is used to establish calls between two H.323 entities
H245: generally transmitted on a separate TCP connections by most older endpoints
REGISTRAR: serverside component that allows SIP REGISTER requests
IEC: international escape code
NEC: national escape code
AC: area code
NUM: phone number
ANI / CLI – Automatic Number Identification or Caller Line Identification
IVR – Interactive Voice Recognition
LCR: least cost routing (price)
BRS: best route selection (price + quality + other settings)
-ANI/CLI authentication:  Automatic Number Identification/Calling Line Identification
-Toll free: is a special telephone number, in that the called party is charged the cost of the calls by the telephone carrier, instead of the calling party. This can be configured as a normal access number (power user) and eventually with higher billing (because we will be the billed party in this case)
-local DID number: normal access numbers. Usually you will have separate DID numbers for different regions to minimize enduser costs
-callback: DID or toll free number configured as power user with iscallback set to the required IVR
-ANI callback: same as callback with User-ID based authorisation (A number)
-Virtual Numbers (DID): "real" phone numbers allocated for users. You have to buy DID numbers from CLEC or any other service provider like
-SMS callback: callback triggered by received SMS message. You have to subscriebe to a two way sms service like users can be authenticated by sender ID, pincode or username/password insterted to the sms text.


Delete old database backup


You can automate backup cleanup by setting the following global config values:
Deldbbackup: days to keep (-1 disables cleenups)
Dbbackupdir: database backup directory
dbdelbackupdir1, dbdelbackupdir1, dbdelbackupdir3:  database backup subdirectories
This feature is useful, when the database engine doesn’t have cleanup feature.


How to set up holiday billing


In the price form in “Time Definitions” select the “Holiday” entry
Set the priority higher in the Directions settings


 How the check your ASR (or ACD, SL, CDRC) for the traffic sender “A” in the last week?


1. In the date-time drop-down list, select the “Last Week” field
2. In the “Select Direction” form set the “Source” (left side) “Type” to traffic sender, and select “A” in the “Name” drop-down list (or type “A” manually)
3. Launch the “Basic Statisitcs” form under Monitoring.
4. Clear the “Group by” option (select  the first  “-“ line)
5. Make sure the ASR checkbox is checked
6. Click on (Re)Load
7. Depending on current server config and current load this query may take some time (on a usual configuration this will take 2 second)


How to treat specific weekends as weekdays?


Set up a new entry in the holidays form and don’t set as holiday (uncheck the checkbox)