help_documentation
Σύνολο εγγραφών στον πίνακα: 1581
| help_id | help_title | help_text |
|---|---|---|
| servers-balance_rank | Balance Rank | <QXZ>This field allows you to set the order in which this server is to be used for balance dialing, if balance dialing is enabled. The server with the highest rank will be used first in placing Balance fill calls. Default is 0.</QXZ> |
| servers-balance_trunks_offlimits | Balance Offlimits | <QXZ>This setting defines the number of trunks to not allow the balance dialing processes to use. For example if you have 40 max trunks and balance offlimits is set to 10 you will only be able to use 30 trunk lines for balance dialing. Default is 0.</QXZ> |
| servers-recording_web_link | Recording Web Link | <QXZ>This setting allows you to override the default of the display of the recording link in the admin web pages. Default is SERVER_IP.</QXZ> |
| servers-alt_server_ip | Alternate Recording Server IP | <QXZ>This setting is where you can put a server IP or other machine name that can be used in place of the server_ip in the links to recordings within the admin web pages. Default is empty.</QXZ> |
| servers-external_server_ip | External Server IP | <QXZ>This setting is where you can put a server IP or other machine name that can be used in place of the server_ip when using a webphone in the agent interface. For this to work you also must have the phones entry set to use the External Server IP. Default is empty.</QXZ> |
| servers-web_socket_url | Web Socket URL | <QXZ>For systems running Asterisk 11 and higher, this is the URL that a WebRTC phone needs to connect to the server.</QXZ> |
| servers-external_web_socket_url | External Web Socket URL | <QXZ>For systems running Asterisk 11 and higher, this is the External URL that a WebRTC phone needs to connect to the server. This External web socket is used if the Phone entry has Use External Server IP set to Y. Default is blank, which will use the normal Web Socket URL.</QXZ> |
| servers-active_asterisk_server | Active Asterisk Server | <QXZ>If Asterisk is not running on this server, or if the dialing processes should not be using this server, or if are only using this server for other scripts like the hopper loading script you would want to set this to N. Default is Y.</QXZ> |
| servers-auto_restart_asterisk | Auto-Restart Asterisk | <QXZ>If Asterisk is running on this server and you want the system to make sure that it will be restarted in the event that it crashes, you might want to consider enabling this setting. If enabled, the system will check every minute to see if Asterisk is running, and if it is not it will attempt to restart it. This process will not run in the first 5 minutes after a system has been up. Default is N.</QXZ> |
| servers-asterisk_temp_no_restart | Temp No-Restart Asterisk | <QXZ>If Auto-Restart Asterisk is enabled on this server, turning on this setting will prevent the auto-restart process from running until after the server is rebooted. Default is N.</QXZ> |
| servers-ara_url | Asterisk Restart URL | <QXZ>If populated, this option will send a URL request for this URL every time that the Asterisk process is auto-restarted on this server. Default is blank for disabled.</QXZ> |
| servers-active_agent_login_server | Active Agent Server | <QXZ>Setting this option to N will prevent agents from being able to log in to this server through the agent screen. This is very useful when using a phone login load balanced setup. Default is Y.</QXZ> |
| servers-generate_conf | Generate conf files | <QXZ>If you would like the system to auto-generate asterisk conf files based upon the phones entries, carrier entries and load balancing setup within the system then set this to Y. Default is Y.</QXZ> |
| servers-rebuild_conf_files | Rebuild conf files | <QXZ>If you want to force a rebuilding of the Asterisk conf files or if any of the phones or carrier entries have changed then this should be set to Y. After the conf files have been generated and Asterisk has been reloaded then this will be changed to N. Default is Y.</QXZ> |
| servers-rebuild_music_on_hold | Rebuild Music On Hold | <QXZ>If you want to force a rebuilding of the music on hold files or if the music on hold entries or server entries have changed then this should be set to Y. After the music on hold files have been synchronized and reloaded then this will be changed to N. Default is Y.</QXZ> |
| servers-sounds_update | Sounds Update | <QXZ>If you want to force a check of the sound files on this server, and the central audio store is enabled as a system setting, then this field will allow the sounds updater to run at the next top of the minute. Any time an audio file is uploaded from the web interface this is automatically set to Y for all servers that have Asterisk active. Default is N.</QXZ> |
| servers-recording_limit | Recording Limit | <QXZ>This field is where you set the maximum number of minutes that a call recording initiated by the system can be. Default is 60 minutes. This setting also limits the amount of time a 3-way call that has been left by an agent will stay up before it is terminated.</QXZ> |
| servers-carrier_logging_active | Carrier Logging Active | <QXZ>This setting allows you to log all hangup return codes for any outbound list dialing calls that you are placing. Default is N.</QXZ> |
| servers-gather_asterisk_output | Gather Asterisk Output | <QXZ>This setting allows you to activate a process that can run every 5 minutes on an active asterisk server and log the SIP/IAX peers and registry output along with the last 1000 lines of Asterisk CLI output. This output is then available to be displayed in the Asterisk Output Report on the Admin Utilities page. Default is N for inactive.</QXZ> |
| servers-conf_qualify | Conf Qualify | <QXZ>This setting allows you to add or remove the qualify entries in the Asterisk conf files. Default is Y for active.</QXZ> |
| servers-custom_dialplan_entry | Custom Dialplan Entry | <QXZ>This field allows you to enter in any dialplan elements that you want for the server, the lines will be added to the default context.</QXZ> |
| servers-active_twin_server_ip | Active Twin Server IP | <QXZ>Some vicidial systems require setting up vicidial servers in pairs. This setting is where you can put the server IP of another vicidial server that this server is twinned with. Default is empty for disabled.</QXZ> |
| conf_templates-template_id | Template ID | <QXZ>This field needs to be at least 2 characters in length and no more than 15 characters in length, no spaces. This is the ID that will be used to identify the conf template throughout the system.</QXZ> |
| conf_templates-template_name | Template Name | <QXZ>This is the descriptive name of the conf file template entry.</QXZ> |
| conf_templates-user_group | Admin User Group | <QXZ>This is the administrative user group for this record, this allows admin viewing of this record restricted by user group. Default is --ALL-- which allows any admin user to view this record.</QXZ> |
| conf_templates-template_contents | Template Contents | <QXZ>This field is where you can enter in the specific settings to be used by all phones and-or carriers that are set to use this conf template. Fields that should NOT be included in this box are: secret, accountcode, account, username and mailbox.</QXZ> |
| server_carriers-carrier_id | Carrier ID | <QXZ>This field needs to be at least 2 characters in length and no more than 15 characters in length, no spaces. This is the ID that will be used to identify the carrier for this specific entry throughout the system.</QXZ> |
| server_carriers-carrier_name | Carrier Name | <QXZ>This is the descriptive name of the carrier entry.</QXZ> |
| server_carriers-carrier_description | Carrier Description | <QXZ>This is put in the comments of the asterisk conf files above the dialplan and account entries. Maximum 255 characters.</QXZ> |
| server_carriers-user_group | Admin User Group | <QXZ>This is the administrative user group for this record, this allows admin viewing of this record restricted by user group. Default is --ALL-- which allows any admin user to view this record.</QXZ> |
| server_carriers-registration_string | Registration String | <QXZ>This field is where you can enter in the exact string needed in the IAX, SIP or PJSIP configuration file to register to the provider. Optional but highly recommended if your carrier allows registration.</QXZ> |
| server_carriers-template_id | Template ID | <QXZ>This optional field allows you to choose a conf file template for this carrier entry.</QXZ> |
| server_carriers-account_entry | Account Entry | <QXZ>This field is used if you have not selected a template to use, and it is where you can enter in the specific account settings to be used for this carrier. If you will be taking in inbound calls from this carrier trunk you might want to set the context=trunkinbound within this field so that you can use the DID handling process within the system.</QXZ> |
| server_carriers-protocol | Protocol | <QXZ>This field allows you to define the protocol to use for the carrier entry. Currently only IAX, SIP and PJSIP are supported.</QXZ> |
| server_carriers-globals_string | Globals String | <QXZ>This optional field allows you to define a global variable to use for the carrier in the dialplan.</QXZ> |
| server_carriers-dialplan_entry | Dialplan Entry | <QXZ>This optional field allows you to define a set of dialplan entries to use for this carrier.</QXZ> |
| server_carriers-server_ip | Server IP | <QXZ>This is the server that this specific carrier record is associated with. If you set this to 0.0.0.0 then this carrier entry will be put on all active asterisk servers.</QXZ> |
| server_carriers-active | Active | <QXZ>This defines whether the carrier will be included in the auto-generated conf files or not.</QXZ> |
| conferences-conf_exten | Conference Number | <QXZ>This field is where you put the meetme conference dialplan number. It is also recommended that the meetme number in meetme.conf matches this number for each entry. This is for the conferences in the astGUIclient user screen and is used for leave-3way-call functionality in the system.</QXZ> |
| conferences-server_ip | Server IP | <QXZ>The menu where you select the Asterisk server that this conference will be on.</QXZ> |
| server_trunks | Server Trunks | <QXZ>Server Trunks allows you to restrict the outgoing lines that are used on this server for campaign dialing on a per-campaign basis. You have the option to reserve a specific number of lines to be used by only one campaign as well as allowing that campaign to run over its reserved lines into whatever lines remain open, as long at the total lines used by the system on this server is less than the Max Trunks setting. Not having any of these records will allow the campaign that dials the line first to have as many lines as it can get under the Max Trunks setting.</QXZ> |
| settings-default_external_server_ip | Default External Server IP | <QXZ>If set to 1, this option will enable you to have your web phone located on a different server</QXZ> |
| settings-use_non_latin | Use Non-Latin | <QXZ>This option allows you to default the web display script to use UTF8 characters and not do any latin-character-family regular expression filtering or display formatting. Default is 0.</QXZ> |
| settings-enable_languages | Enable Languages | <QXZ>This setting allows you to enable non-English language translations on the system. A new section called Languages under the Admin section will also be available to manager Languages. Default is 0 for disabled.</QXZ> |
| settings-language_method | Language Method | <QXZ>This setting defines how the language translation works when pages are loaded. The MYSQL method performs a database query for every display of a phrase in the interface. The DISABLED method will always display -default English- no matter what other settings are set to. Default is DISABLED.</QXZ> |
| settings-default_language | Default Language | <QXZ>This is the language that the agent and administrative interface will default to before the agent logs in. Default is -default English-.</QXZ> |
| settings-webroot_writable | Webroot Writable | <QXZ>This setting allows you to define whether temp files and authentication files should be placed in the webroot on your web server. Default is 1.</QXZ> |
| settings-agent_disable | Agent Disable Display | <QXZ>This field is used to select when to show an agent notices when their session has been disabled by the system, a manager action or by an external measure. The NOT_ACTIVE setting will disable the message on the agents screen. The LIVE_AGENT setting will only display the disabled message when the agents auto_calls record has been removed, such as during a force logout or emergency logout. Default is ALL.</QXZ> |
| settings-frozen_server_call_clear | Clear Frozen Calls | <QXZ>This option can enable the ability for the general Reports page and the optional AST_timecheck.pl script to clear out the auto_calls entries for a frozen server so they do not affect call routing. Default is 0 for disabled.</QXZ> |
| settings-allowed_sip_stacks | Allowed SIP Stacks | <QXZ>This option will determine which SIP modules in Asterisk will be allowed to be configured on your dialers for Phones and Carriers. The standard up until Asterisk 16 is -SIP- which uses the older chan_sip module. The newer PJSIP can be used with Asterisk 16 and higher, if configured properly. Default is SIP.</QXZ> |