Overall System Settings
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General
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The "System Name" is used in several places to identity the PBX. For example, when the PBX sends out emails that contain performance information about the PBX, it uses this name in the subject line.
The "Audio Language", the "Tones", the "Web Language" and the "Timezone" determine which language is being used for the speech, the tones and the web interface. For more information, see Localization.
Administrator Login
It is very important that the login as system administrator is protected. Therefore, you should set a reasonable safe password for the administrator login.
By default, the username is "admin" and the password is "Biz8000".
If you forgot your password, please contact your dealer or Bizfon Customer Service.
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Appearance
The "Default CDR listing" size tells the system how many CDR records to display on the web interface. This setting should avoid that the user gets too many CDR on the display at a time.
The "Keep CDR Duration" setting defines how long the CDR are kept in the database. By default, this setting is 14 days. The duration is expressed in time unit. A time unit may seconds (put a 's' behind the number), minutes (put an 'm' behind the number), hours (put a 'h' behind the number) or days (put a 'd' behind the number). An example would be "10d", meaning that the CDR are kept for ten days.
The "SOAP Trusted IP" and the "SOAP CDR URL" are only available if the license key contains a SOAP flag. See the Simple CDR Format if you are connecting to billing tools that use a simple IP-based format.
Most SIP phones do not have a recording button. In order to have the recording feature also available for those devices, you may define DTMF keys that start and stop recording. The keys must be one character and can be 0-9, * and #. Recording is triggered only on connected calls. Please be aware that the other side may hear the tone and that this feature might have side effects on other features, for example when you are calling an external mailbox and use the keys for navigating. The "Recording Location" defines where calls are being recorded. For more information about Recording, see the page Recording.
Performance
The "Maximum Number of Calls" setting defines how many calls the system allows at the same time. Because every call takes a certain portion of the available CPU, allowing too many calls will affect the quality of all ongoing calls. By limiting the number of calls on the CPU, you can reject calls that would otherwise potentially degrade the overall performance.
The "Maximum Duration of Call Recording" sets an upper limit to call recordings. This setting is important because recording files might become very large and can cause problems with the system performance. There is another setting that limits the recording of a mailbox message, which is a domain setting.
In a SIP environment, the registrar determines how long a user agent may be registered. Short registration times have a negative impact on the performance, but make sure that the user agents stabilize quickly after they lost connection to the PBX. The "Minimum Registration Time" and the "Maximum Registration Time" settings are used to define the lower and upper limit for the registration time. Typical values are in the range of a few minutes up to several hours. The settings use seconds as the unit.
If the registering user agent is behind NAT, the PBX uses the settings "UDP NAT Refresh" and "TCP/TLS NAT Refresh". The PBX registers agents that use the UDP transport layer only for a short time, so that the user agents will reregister quickly and keep the NAT bindings alive this way. Typically the settings for UDP should be in the range from 15 to 45 seconds, while TCP/TLS connection don't need to refresh the bindings so often, a value of a few minutes are ok in most situations.
The "Maximum call duration" settings set the upper limit for the call duration. By default the setting is two hours, but you might make it longer if you have long phone calls. This setting is good to keep your call list clean, for example if one mailbox talks to another mailbox.
More information about the registration procedure can be found in Registration Duration.
SIP Settings
SIP specifies for certain headers a short form. Short headers have the advantage to save some space in the messages, which reduce the overall probability that you run into problems with maximum message size in UDP. Although it is very simple to support this, some devices are not able to deal with the short headers. Therefore, the PBX offers both short and long headers. In order to maximize interoperability, the default value for Use Short SIP Headers is long; if you are running into UDP packet fragmentation problems (message size above 1492 bytes), you should switch to the short header form.
SIP also has it's own multicast group. Usually a SIP device knows where to send the requests; however during boot-up and configuration, a user-agent might want to locate the PBX with a multicast request. Therefore, the PBX offers the setting Listen to sip.mcast.net. If this setting is turned on and you are using user-agents with the multicast detection feature, you can just plug the devices into the network and they will get their configuration information automatically.
In hosted environments, the service provider might want to set the trunks up and hide this feature from his customers. If Allow domain admin to change trunks is set to "no", then the domain administrators can see their trunks only in the dial plan, but are not able to make changes to the trunks.
When the PBX starts a call, that same call may come back to the PBX, creating a loop. This is a dangerous situation, because it might initiate the same call again and again, ending up in many calls that take a lot of resources. Therefore, the PBX must detect such a loop. In environments where an external SIP proxy routes the call from one PBX domain to another, a simple loop back detection based on the call-id is too pessimistic. Therefore, in such environments you might want to allow such calls and turn the loop back detection off.
When a user presses a key on the telephone, the PBX must be able to understand that key press. In telephony system, this mechanism is typically called DTMF (see http://en.wikipedia.org/wiki/DTMF). In VoIP, DTMF should usually be sent via the out-of-band method (RFC2833), which makes it easy and failsafe for the PBX to detect those tones. However, there are sometime devices, which are not supporting this method. In this case, the PBX must decode and analyze the media stream and perform this detection. This is erroneous and costs additional CPU performance. It is strongly recommended not to use this feature and to replace devices which do not support out-of-band with devices that do.
In environments where the service provider controls the PBX from a centralized location, the setting "Remote SIP management" is used to allow the provider to send commands to the PBX (for example, for re-reading the configuration). By default this setting is off, but if you are using such an environment this setting needs to be turned on.
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