This menu, located at the Settings/System/General, contains various settings stored in the database and checked on the voipswitch application start. The menu is divided into four tabs:
- Main settings
- Failed calls
- Rerouting and ending calls
- Active calls recording
The options grouped under the first tab are explained below. They are global settings related to call processing in various stages.
Limit ring time – sets the maximum ring time expressed in seconds after which a call will be either rerouted to another destination (according to the Routing plan) or Answering Rules will be triggered or, if no other route is available, the call will be dropped. The default value is 600 sec. The time is counted from the INVITE request and stops on receiving 200 OK.
Limit call duration - defines the maximum call duration expressed in minutes after which the call will be dropped by the softswitch. It can prevent some random situations when the call hasn't been finished by either side for long time.
Use media timeout – when this option is on, the softswitch controls the media flow and will drop the call if either of the calling parties stops sending RTP packets for defined time. It works only for calls where RTP packets go through the server (full proxy mode). For calls established with the use of ICE mechanism the settings is ignored. Also when a party put the other part on HOLD the softswitch will not trigger the timer.
Limit number of hops - this option limits the number of hops made if the call can't be connected to consecutive destinations with the same prefix and different priorities
Max number of all hops – similar as above with the difference that the limit refers to all failover hops, inclusive of the entries with the same prefixes and those with less matching prefixes, for example when call to prefix 48 fails over to the route with more general prefix 4.
Max number of calls per second – max number of INVITE requests received in one second, all attempts above this limit will be dropped.
Refuse connection if destination tariff is higher than client rate - prevents connecting calls if the tariff associated with the destination (cost tariff) has a higher rate for the dialed number than the client's tariff. This option can prevent from incurring losses in case of wrongly defined rates.
Guest account - all calls which come from nonexistent logins will be identified as the client selected in this option. It is used in services where you want to allow access for SIP endpoints which are not your clients. If you group all such calls under one account it will be then easier to control and limit such traffic
Authenticate by IP address – eligible only for wholesale type of clients. When this option is on, the switch will be trying to authenticate incoming requests by IP address first by checking if the address belongs to a wholesale client.
Use web passwords – makes the system use separate passwords for SIP and separate for web/mobile clients’ access (log in to the self-care portal or to the softphones)
Alerting timeout – max time from the moment the INVITE has been sent until the 180 Ringing or 183 Session progress response is received. If not received the call will be ended by the softswitch.
Use time spans in the routing plan - enables the time spans feature in routing
Use load sharing - enables load sharing in the routing plan, causing the softswitch to distribute the traffic to multiple destinations specified in the appropriate routing entry, more explained in the Routing chapter
Use tariff plans - enables the tariff plans feature;
The set of options in this tab determines whether some special types of failed calls should be written to the database. The types are related mainly to the global settings or particular restrictions. The latter are related to per account or per dialed number admission limits.
You can select which of the failure types should be logged in the database:
There was no destination in the dialing plan – if the dialed number is blocked on the routing level,
Client had no money – call failed because the cost of the minimal billing step was higher than the account balance.
Destination was offline – voipswitch sent an INVITE request to the remote endpoint but has not received any response until the timeout elapsed.
Limit was reached – the account limit of concurrent calls has been reached.
Dialed number was disabled – blocked in the routing plan by explicitly disabling the best matching prefix (or whole number)
There was no tariff – blocked on the billing level, there is no matching prefix for the dialed number in the tariff assigned to the calling party account, for example if you want to exclude certain destination from the service
Source and destination do not have same codecs – rejected because there is no destination endpoint which has at least one common codec with the calling client and the transcoding is not enabled or what is less likely the transcoding server does not support any of the codecs supported by either side
Dialed number has wrong length – the call was rejected because of the limit to the dialed number’s length configured in the Dialing rules either in the client account or in the Routing plan.
Rerouting and end reasons
Re-routing is a function that directs calls through another route if the route which is the best match for the called number or, in the case if there are several entries with the same prefix, the one with the highest priority, cannot connect the call.
By default voipswitch will be re-routing all failing calls no matter what error code is received from the terminating endpoint. Re-routing section allows you to specify which particular events should make the softswitch proceed to the next route. You can also define in which cases the re-routing must not be triggered.
To add a new rule select an end call reason from the drop down list in the Re-routing calls area. Then, by clicking on appropriate checkbox choose whether the end reason should cause that the call is rerouted or not. After pressing the add button the new rule will appear in the list above the rectangle area. At any moment you can remove particular rule from the list.
End reasons mapping
End reasons mapping section allows for replacing the end reasons generated by voipswitch or received from the called endpoint to any other reason which will be sent to the originating client.
Voipswitch platform’s related end reasons have to be translated to adequate SIP responses which will be sent to the client. The reasons are listed on the left side and the corresponding SIP responses on the right. By default the softswitch uses a set of predefined codes, if you wish to change any of them click on the drop down list and choose the more suiting response. The platform ending reasons include:
Unauthorized call – the call failed on one of the authentication/authorization functions; by default translated to 403.
No money to make a call - the client has run out of money on his account; by default translated to 402.
Number doesn't exist in the destination tariff – there is no matching prefix in the cost tariff assigned to the destination. This option works with the gateway type of destination route. By default translated to 404.
Number has wrong length - dialed number length doesn't match rules set in the routing plan prefix. For example if you want that the number should consists of exactly 4 characters, which is denoted by , and the dialed number is of a different length, i.e. is longer or shorter than 4 characters; by default translated to 484.
Number doesn't exist in tariff – there is no matching prefix in the tariff assigned to the client account; by default translated to 404.
Number is disabled in tariff – the dialed number is disabled either in client or destination tariff; by default translated to 404.
Calls limit – calls limit for a given account has been reached; by default translated to 503.
Destination offline - destination which voipswitch is trying to connect to is not responding; by default translated to 480.
Ring time occurred - ring time limit has been reached; by default translated to 480.
Codec problem - codec misconfiguration problem, client is using a codec which is not allowed in his account; by default translated to 488.
Unknown reason - the disconnect reason is unknown; by default translated to 404.
The SIP responses in the drop down lists are from the below ranges:
- 4xx—Client Failure Responses
- 5xx—Server Failure Responses
- 6xx—Global Failure Responses
SIP end reasons mapping
In addition to replacing the internal failure codes the softswitch is also capable of replacing the SIP responses received by the destination endpoint. In effect the client will get a different response according to the mapping defined in this section.
To set the mapping simply choose the desired response code from the upper drop down list in the Gateway end reasons area, then pick the response code from the drop down list below and click add to create a new mapping rule. The added rules will be shown in the list above.
Active calls log
The active calls term refers to connections which are pending on the softswitch. They can be at different state such as:
When the call ends, regardless if it has progressed thru all the states or was ended earlier (failed) the softswitch writes the call’s details to the database. The active calls states are stored in the application memory.
In some cases however you may want to share the information about the active calls with other applications, for example with Callshop, VUC or a 3rd party application. When you check the write to database option it will make the softswitch insert new record for each new call and update the record when the call changes its state. Moreover, you can decide on which of the call states the update should take place. Active calls are stored in the currentcalls table in voipswitch schema.