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VoipSwitch - a VoIP software developer; its main product is a Class 5 softswitch, mobile dialers, Rich Communication Suite and OTT complete platform. View and Download Quintum Tenor AF product manual online. VoIP Switch. Tenor AF Switch pdf manual download. PBX - Unified Communication · RCS. DIDs (Direct Inward Dialing). FAQ VoipSwitch Frequently Asked Questions. Gateway WholeSale Training 1. Gateway WholeSale Training Video Manual.
In "proxy all" VoipSwitch does not allow codec negotiation directly between endpoints and instead will negotiate itself with each endpoint in part. While in "proxy only signaling" the endpoints can negotiate directly the codecs, it is possible to choose any codec that both endpoints support, even those that are not listed in VoipSwitch settings. IP numbers are the list with authorized IP addresses. Cost of calls coming from ip assigned to a client is taken from his account.
You can set here an unlimited number of addresses, but an IP can be enabled only for one GWClient at a time. Under the IP numbers list there is a field where to write the new addresses to be added in the list. Use the Add IP button after you fill it. It is possible to add ip addresses in range. After clicking Add Range button the dialog as on screen below will appear.
There it should be set starting ip and ending ip. VoipSwitch will use them as boundaries to create appropriate entries in ip numbers list. For starting ip will be added 1 till it reaches ending ip. Connect client immediately Enable this only when all calls of a client do not connect to any destination. This will open the media channel immediately after routing but in most cases will generate also false billing because the calls will be declared answered immediately. So this feature is for extreme cases only.
2.1 Configuracion Inicial
Do not use it for normal users. Calls limit Used to a limited number of concurrent calls being send from gateway. When number of calls is equal to this limit any new calls from this client will be rejected. This is also checked for calls in progress and connected apiece. PC2Phone clients This type of client is for pc2phone dialer and web2phone page access only.
Pc2phone is a proprietary application that allows clients who have a valid pc2phone client account to connect to the VoipSwitch and initiate and also to receive calls.
This dialer uses particular communication ports and is not compatible with other systems. The settings for pc2phone clients are very simple and the fields have the same meanings as for GWclients.
Pc2phone application always uses g PC2phone client is allowed to make only one call at the same time. This type of client is hot billed Login and password defined for every client are used to log using pc2phone application More about PC2Phone application is described here.
Setting termination on PC2Phone client is described in dialing plan section of manual. GK clients This client type is used for those devices behind NAT, or those that change the IP often or simply want to register with a user and pass only. The client will have to configure his device to register to VoipSwitch's Gatekeeper when using h protocol or Registrar for SIP protocol using the user and password he received.
Also he will need to enter in his device configuration the IP of Voipswitch and the Gatekeeper name that is by default Gatekeeper in case he uses h Parameters supported codecs, calls limit have the same meaning as desribed for GW clients. Login and password are used to log from device to VoipSwitch acting as Registrar or Gatekeeper.
VoipSwitch will recognize automatically protocol being used to log. Clients of this type are hot billed but only when one port of device uses one login and password. Otherwise hot billing the function may not work, for example when one call is started and then second port using the same account calls.
The system will not be able to calculate properly remaining account state, and account balance can go below zero.
To eliminate this possibility calls limit value should be set to 1. After sucessfull loging to VoipSwitch a device will appear in Registered clients. Different icon will be used for h and sip devices. Detailed description can be found in manual for callback system. Every connection made by callback client consist of 2 calls and both are charged.
Main callback features are listed below: After being connected to destination number a client can finish call and pick another number without disconnecting source leg of connection.
After setting appropriate scenario a client can hear account state and remaining time announcement after every call made. Checking it will block redirection for pc2phone.
VoipSwitch owner can have clear dialing plan with real country codes. If destination is IP phone logged to VoipSwitch as GK Registrar client than in most cases it will respond only to the number being the same as login. If we want to redirect some DID number then rule must be defined as show on screen. Every device has special field used as number to which it responds.
When creating this Dialingplan rule it must be set this distinct number Telephone number and any Destination because it will not count in the routing process. Cisco ATA. Using these rules it can be modified. PC2Phone or Common client account that will have the Login as the dialed number..
Option disable "folow me" for DNIS mapping is used to disable follow me for entries with any map to dnis option. Some carriers require it for authorization or different billing.
For some reason expected gateway inaccessibility we can disable prefix without removing it from the dialing plan. Client can define on web page that depending on specified reasons call should be redirected. Such prefix will not be used to send calls and later it can be easily restored. This scenario is playing recording as no answer voice. Even if there are other prefixes matching the dialed number rerouting will not be made.
DID number used to activate callback can be set in dialing to Voipbox and Play file scenario. Instead of phone number a client can define that it should go to voicemail. Voice is played to client but connect is not returned so there is no charging for such call. Common practice is to block special expensive numbers. When this option is used Follow me feature is automatically turned on.
It means that a call can follow if a client is not answering. GK or common clients. Different destinations can be used in different time. For some service numbers account state information. Not all devices can generate fake ring. But Cisco ATA and others can. It is used commonly with callback service. After choosing dialing plan position from a menu tree the import export buttons will appear in the upper right corner of the screen.
Export buttons for exporting dialing plan positions Clicking on Export button causes exporting all dialing plan positions. System will ask then about location and name of export file and then it will be completed.
Such file can be opened in Notepad or even by Excel for further modifications. If filter is applied only filtered records will be exported. Format used for export is coma delimited CSV files. It is not recommended to modify it manually Depending on which protocol is using specified route SIP or H this value is differently decoded. Tech prefix stores value for part defined in VSM as Rules for modifying client's data. But there is a feature which allows importing incomplete rows from file.
Don't jump.. This text value has coded conversion rules for the dialed number. The only one required field is telephone number. Type has coded definition of values defined in VPSconfig as Special properties. It shouldn't be modified directly but rather copied from existing row. It is quite complicated to manipulate directly those values but anyone interested can define some test entry with valid conversion and later use it in these files for import purpose. Examples of string manipulations are defined here Call type value is binary coded and it defines dialing plan mode.
If other fields are empty for example ''.
This form is the same as the one used with adding or editing dialing plan positions. Some parts are hidden or displayed depending which fields are in import file. Only setting two comas without any character between them will cause asking about missing value. Filling missing information in dialing plan import Only first row is checked and other rows are imported with values filled using this form. Space character or two apostrophies '' between comas won't be taken as empty.
These values must be picked up in a form.. Example of such incomplete row ''. This row will cause the system to ask about destination device and call type. SIP proxy or gatekeeper to calculate cost of termination reseller of any level to calculate cost for him for special usage like Tariff to DNIS or Tariff to ANI Every tariff is defined the same way and only assigning them causes different usage.
Every tariff is built of prefixes with assigned minute price for them. Few parameters can be defined to whole tariff and some of them can be defined for specific prefixes.
A call will be connected only if the prefix of the dialed number exists in the tariff. Prefix must exists in tariff assigned to a client. All tariffs are defined in one place and later a tariff can be used for different purpose.
All the dialed numbers without matching prefixes in tariff table will be rejected. Tariff can be assigned to: For example when it is set with 6 it means that every 6 seconds one of tenth minute price will be added to cost. Resolution Every rate assigned to prefix is for one minute. Tariff definition allows to charge clients for shorter periods from 1 second to any number of minutes.
Value of resolution is in seconds and defines when part of a minute price should be added to cost of a call. Minimal time Document generated by Confluence on Feb Parameters for tariffs Every tariff is identfied by name and set of parameters used to calculate cost of calls.
After you click it. As resolution it can be set at any value but for clear calculation it should have a value which divides 60 without rest. Resolution is a parameter which is used for that. Surcharge amount Amount added to every call for begining of each call is defined by surcharge time. Tariff multiplier Client can see his tariff rates and prefixes on the web page after logging. Value 0. Even if a call is connected and disconnected after few seconds a client will be charged for this time.
Tariff prefixes Document generated by Confluence on Feb Surcharge time This value defines time in seconds which can be charged by surcharge amount. Until a client recalculates cost of a call manually he will not be aware of it. Unchecking this option makes setting tariff easier and will speed up cost calculation.
For example if the Surcharge time field is seconds and Surcharge amount is 0. For example when surcharge time is defined as 10 and duration of call was 60 then surcharge amount will be added to cost of a call and the rest of 50 seconds will be calculated using minimal time and resolution.
Rate multiplier is changing cost of every rate by multiplying it by this value. Time span When this option is checked it allows to define different rates used for different days or hours. When surcharge amount is set at 0 then time defined in surcharge time will be free for client. This value is used to define minimal time for which a client will be charged after connection.
Resolution steps required to cover minimal time will be used to calculate cost. Tariff addition Similar to tariff multiplier but instead of multiplying it adds some value to every rate. Remaining time of a call will be charged using other rules described in minimal time and resolution.
Insted of this add two rules first one: Using this option allows rates visible by a client to be changed during call. If you would like to setup some rate for hours crossing the midnight ie 8. If surcharge time is set at 0 then it will work as connection fee for any connected call. If it is set 1. After the Surcharge time expires the billing will start as if it is the beginning of the call.
Prefixes defined for a tariff are presented on screen after clicking tariff name. Having less number of prefixes in tariffs can speed up processing. The same description can be used for different prefixes and later can be used for more general grouping.
If price is the same for many similar prefixes it must be considered if it cannot be replaced with one more general prefix.
Calls longer then this time will be billed normally with the whole duration including this grace period. If in tariff are defined prefixes 4. If one of them has different rate then it can be set 48 with 0. It allows to set different value for specific prefixes. For example and are Poland mobiles however with different cost. Descriptions defined in every prefix should be filled properly because they are used later in detailed billing.
Examples Client calls number 48 Grace period If any connected call is shorter then this time it will appear in CDR with 0 cost. Common practice for setting a tariff for given country is to define general number for a country and then only define more detailed prefixes with different rates. The same as in dialing plan longer more detailed prefixes matching dialed number are taken first before shorter.
Tariff can have any number of prefixes defined in it. Examples Having prefix Some parameters are the same as for tariff. For every prefix some parameters can be set to modify calculation of cost. If rate for and 48 are the same then can be removed. If such parameter has value higher then 0 it will be used instead of the one defined in tariff.
Later in summary we can see it grouped by description and see how many calls went to polish mobiles. Tariffs with smaller number of prefixes are easier to manage and with higher traffic can be processed faster. Prefixes are assigned with rates and descriptions. Minimal time See minimal time defined for tariff. Rate addition See rate addition defined for tariff Disable prefix Any number which matches this prefix will be blocked from processing.
Minimum Time. Similar to option used in dialing plan used to block some prefixes. To Day. The file should look like this: Grace Period. Rate multiplier See rate multiplier defined for tariff. Resolution See resolution defined for tariff.
System will reject such call. For this you will have to prepare the file in the following specific format order of columns: Resolution Be sure you don't have column names in the text file. Below is an example of how it should be set to have offpeak and onpeak rates. The file should not contain comments or column headers and data should start from the first row. From Hour.
To Hour. It is possible to define from which day of a week to which day and between which hours this prefix will be valid. From day. From Day. When you have the file ready. VoipSwitch Manager will stop the importing process and announce the error if the process fails.
You will be asked to select the text file. The records from the file will be added to the existing records in the tariff. Some fields can be empty but if you won't. You must remove the existing records before importing if you want to replace them. There is a Remove all button that will delete all rates in that tariff for your convenience. Changing tariff for clients Client can have one tariff assigned to him but there are a few ways to change such tariff depending on information sent to VoipSwitch.
Detailed description how to set it is described here. One number used to call to IVR system can be toll free and other charged for every connection. It can be used to differentiate cost of calls when client is calling from abroad.
This function is named as 'Tariff to ANI" and is used only while providing the 'calling cards' service. This function is exceptionaly usefull in the USA. Using NPA function This function allows a user to change the tariff depending on the number that is dialled and the number that makes the call using Caller ID.
In the Others section one can find a Numbers Table with a location name linked to it. A simple comparison of numbers is not enough. Only by using this feature we can differentiate tariff used to charge the same client. Tariff to ANI function is changing tariff assigned to a client to some other depending on caller id coming to VoipSwitch.
Every reseller can set different rules for his clients on how tariff should be changed depending on the number called. Typical scenario is to set different number for calling cards and then depending on different tariffs it will be used to charge clients.
Procedure of tariff change. It is used in calling cards service. Option used to compare 2 tariffs. The user calls a number that begins with a certain string of numbers. Matching prefixes are listed in output window and can be modified. Mistake in one rate can cause big loss for VoipSwitch owner so it is very important to check every tariff before assigning to clients.
It is possible to define how much bigger or smaller it should be. After finding the location the system compares the beginning of the clients Caller ID and if it is assigned to the same location it uses the Intrastate Tariff. If one tariff should be lower. Working with traffic and clients requires many tariffs. It is possible to chose two tariffs and define criteria for comparing tariffs.
The system checks which localisation has such a number assigned. After clicking it one can configure the Tariffs for the client in a new window. If there is no match the tariff used is Interstate.
Tariff comparer is one of features available in VSM application. Tariff comparison is more complex Document generated by Confluence on Feb Example Tariff comparer Option is available under tariff node. Comparing tariff used for clients and used to calculate cost can find mistakes and avoid money loss if it will be noticed later after some calls.
Tariff assigned to resellers should be also compared with cost tariff to earn on every prefix. Value from Tariff 2 voice rate is used to multiply prefixes from Tariff 2. Tariff comparer should be used every time when prefixes are changed.
OpenSS7 Carrier VoIP Switch
Make sure all prefixes assigned to reseller tariff are higher than in cost tariff. If the prefix in tariff is not the same the best matching entry is taken to comparison. Examples To find prefixes from Tariff 1 which are bigger than in Tariff 2. The cost of such a connection depends on the type of the incoming call: Then the connection is transferred to the IVR system. In order to configure the service 'tariff to ANI' correctly. This function is used mainly while providing the 'calling cards' service.
In order to use this function effectively. Typical applications The function 'Tariff to ANI' enables to change the tariff for a client according to the telephone number from which the connection is being made. Within the frames of this example. For instance in Poland the callers who make a phone call from a mobile phone have their Caller ID starting with 60 or Other numbers.
The phone calls coming to VoipSwitch from the telephone network are made by authorizing the telephone operator's gate.
Having saved changes for such a client, it is possible to make connections and the appropriate tariff will be used according to the Caller ID.
In the following example you can see, that one IVR Client login called three times number , but in every case with other Caller ID. Reported problems Clients calling VoipSwitch can use different gateways. Some of them are sending different Caller ID format. One operator sends the Caller ID in the above-mentioned format, and the other sends the Caller ID with the country code 48 at the beginning, hence the format is Rules on how to make it are described here.
ANI Tariff used by resellers. Resellers cannot define their own tariff to ANI rules, but may use the ones defined globally. To do so every client added has to have "Tariff to ANI" option marked see picture below.
Enter to the Edit client page, mark "Choose tariff according to" option, then click Rules button. It is enough to define just the initial digits of the dialled number and to assign the new tariff to them.
Tariff to DNIS This function makes it possible to change the tariff which is used to calculate the cost for a client according to the dialled target number. After ticking that checkbox and confirming the change. Each client has a checkbox in his definition.. Click here for example.. This function may be applied for every type of clients. Each entry will work for a different number dialled by the client. In the first case the client will be charged at a higher rate in order to cover the cost of the toll-free connection from VoipSwitch.
The picture above shows how two entries have been defined. The other one is a toll number One number is free of charge for the client. Those are example calls made by this client. Second call was made to normal access number.
In this example two calls were made to VoipSwich. It is possible to filter calls by: Any call can be traced using date. This section describes every such feature and is divided into subsections listed below. It is possible to see summary reports with charts about gateways usage. Working with filters.
Statistics can help to choose best destinations and modify dialing plan according to such knowledge. Browsing payments made by client help solve misunderstandings. There are helper periods available as Today. Last week. Average duration. Time shift This new option allows to show result using time shift. You may choose columns to export if it suits your needs.
It is useful when you want to compare calls being made to VoipSwitch with the same calls received from carrier and they are not in the same time zone. Avarage revenue. Below grid with calls summaries are calculated for the filtered set of calls.
Total cost. Total duration. After clicking Apply button the list of calls will be refreshed with filtering conditions.
After choosing export file the name dialog as shown on Fig. Informations available in summary section are: Choose Document generated by Confluence on Feb It is coma delimited and can be easily opened in Excel application or in Notepad. Other filtering fileds are the same. Only by using specific type of client it is possible to filter calls for a specific client. Resellers calls Special part of calls is availalble for resellers. It is possible to see calls made by every reseller on every level.
Client login name must be put exact. Filtering by date is also available as shown on Fig. In the form below additional fields are: It is possible to group and filter depending on different criteria.
Values there are similar as in the list but for IE error and reason there are explanation of errors which can be useful for finding error on destination gateway. Average Count Document generated by Confluence on Feb For grouped record it can calculate: Below please find list example reports: External gateways. Grouping by country makes it possible to see as pie chart countries chosen by clients. Choosing many groupings can result in high CPU usage and long operation. Below is a list of possible grouping: Clients type.
Grouping allows to see sums for chosen client. Using such report can be helpful to see clients calling best. By country. Visualization of this report is available in pie chart. Defined criteria can be used as one or as many. Grouping records will be applied for records filtered. For every reseller level it can be presented on a pie chart how big the usage of every reseller in total Document generated by Confluence on Feb Res 1. To see all clients informations specific client login cannot be chosen.
Resellers of any level. Available filters are: It is better to limit time for which such report is being generated and use grouping with caution. External gateway External gatekeeper - Payments Section used to browse payments made by clients. Only visible fields will be exported to file. Values available are: Statistics This section is using calls made and failed to calculate statistics. GW clients. This format can be opened using Excel. Export Every generated report can be exported to CSV file format.
Payments can be checked for filtered date.
They are useful to check quality of gateways. SIP listeners. SMS Callback listeners. Mail settings. Ending calls. Call settings. After clicking Settings node in VSM application it will expand and show different sections as shown on figure. Other settings. Other ports. Callshop listener. Rerouting calls. After changing most settings. Callback listeners. H listeners. Normally it is unlimited. Settings defined there are used mosty by VoipSwitch. If not sure ask VoipSwitch support. If any of them is not used it should be unchecked and it can improve VoipSwitch effciency.
When it is set calls will be authorized and billed for this client. Parameters here are divided into 5 sections: After executing this command new settings should be applied to working VoipSwitch. It is useful to set some value there to avoid endless calling because then calls redirection will not happen. Any change made here should be done carefully and with full awareness of possible effects.
It will work only in full proxy mode when media packets are coming through VoipSwitch. Authorization Other Parameters set in this section turn on and off functions available with VoipSwitch. This parameter allows to limit it. Without this option set with login name all these calls would be rejected. Any option changed before it will be used by VoipSwitch must be saved and settings must be reloaded.
If there are matching prefixes defined in dialing plan. It will also not substract any value from their account. Rerouting calls Rerouting calls allows to find different gateways for dialed number if for any reason a call cannot be connected.
Ending calls Document generated by Confluence on Feb Good example is error "User busy". VoipSwitch is set by default to reroute all calls no matter what error code is received from termination gateway. This message is returned when dialed number is busy. There is no sense in finding other gateway because it will still be busy. It can cause some problems if unchecked by mistake because all calls made by resellers clients won't be assigned to resellers.
Sometimes it can be required to stop rerouting calls so calls can be treated as failed faster then waiting for trying many gateways. Rules set here are applied to all numbers defined in dialing plan. Using this dialog can be an added option allowing not to reroute when such error will be received on VoipSwitch. Similar rules can be added for every error returned by gateway. One is Gateway end reason and number set there will be replaced with End reason sent to client.
Adding error number replacing rules doesn't require restarting VoipSwitch. Possible reasons which could happen in VoipSwitch are: Second part of these settings is for changing error numbers received from destination and passed to clients.
There are two buttons which allow to chose error number. Default behaviour is to pass it unchanged but using these settings it is possible to replace one error number with another. It was built in this way to avoid counting such calls for general statistics like for example ASR. Save failed calls when Calls failing because of VoipSwitch misconfiguration are not stored by default in database.
It is important to set it for both protocols for every reason.
After clicking button rule it will be copied to text box on the right and for next calls matching Gateway end reason it will be replaced. Sometimes in order to have more extended information about such calls it is possible to turn on saving them in failed calls. VoipSwitch denies call processing because of unauthorized request.
IP not added. H listeners Listeners are needed for clients to connect to server. Newly added port will be listed above H listener ports list shown on Fig. There is one IP address H listener has default port You may setup H listeners in section shown on Fig. Every error presented on screen above can be checked to be saved in database in general failed calls table. The length of the history may be different from VoIP switch to VoIP switch and depends on the length of log files and amount of logging events.
The "Support Log" is tenant sensitive. This means a supporter of tenant A is not able to see events of tenant B! The interpretation of a "Support Log" is quite easy and straight forward. With a little experience one will be soon familiar with the interpretation. The "Trace" has a limited history.
The interpretation of a "Trace" PCAP formatted file has to be done in an external application like Wireshark network protocol analyzer. Wireshark offers deep and rich VoIP analysis. The "Trace" is not tenant sensitive.
This means a supporter of tenant A is able to see signaling messages of tenant B!
Due to this open display of information it may be possible that the "Trace" is not available for the supporters and operators on a multi tenant VoIP Switch.
Extended filters enable the supporter to search for specific calls. The filters can be combined with logical AND. The "Call Data" has a limited history.Always send will fill the prefix field in the format! It also provides for continuity test tone generation, detection and loop back.
Supported codecs accepted by remote side. Every other calls will not be visible on the list, but of course this will not influence other calls. Context men will appear with command "Add to reseller".
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