help_documentation
Σύνολο εγγραφών στον πίνακα: 1581
| help_id | help_title | help_text |
|---|---|---|
| phones-DBX_database | DBX Database | <QXZ>The MySQL database that this user should be connecting to. Default is asterisk.</QXZ> |
| phones-DBX_user | DBX User | <QXZ>The MySQL user login that this user should be using when connecting. Default is cron.</QXZ> |
| phones-DBX_pass | DBX Pass | <QXZ>The MySQL user password that this user should be using when connecting. Default is 1234.</QXZ> |
| phones-DBX_port | DBX Port | <QXZ>The MySQL TCP port that this user should be using when connecting. Default is 3306.</QXZ> |
| phones-DBY_server | DBY Server | <QXZ>The MySQL database server that this user should be connecting to. Secondary server, not used currently.</QXZ> |
| phones-DBY_database | DBY Database | <QXZ>The MySQL database that this user should be connecting to. Default is asterisk. Secondary server, not used currently.</QXZ> |
| phones-DBY_user | DBY User | <QXZ>The MySQL user login that this user should be using when connecting. Default is cron. Secondary server, not used currently.</QXZ> |
| phones-DBY_pass | DBY Pass | <QXZ>The MySQL user password that this user should be using when connecting. Default is 1234. Secondary server, not used currently.</QXZ> |
| phones-DBY_port | DBY Port | <QXZ>The MySQL TCP port that this user should be using when connecting. Default is 3306. Secondary server, not used currently.</QXZ> |
| phones-alias_id | Alias ID | <QXZ>The ID of the alias used to allow for phone load balanced logins. no spaces or other special characters allowed. Must be between 2 and 20 characters in length.</QXZ> |
| phones-alias_name | Alias Name | <QXZ>The name used to describe a phones alias, Must be between 2 and 50 characters in length.</QXZ> |
| phones-logins_list | Phones Logins List | <QXZ>The comma separated list of phone logins used when an agent logs in using phone load balanced logins. The Agent application will find the active server with the fewest agents logged into it and place a call from that server to the agent upon login.</QXZ> |
| phones-template_id | Template ID | <QXZ>This is the conf file template ID that this phone entry will use for its Asterisk settings. Default is --NONE--.</QXZ> |
| phones-conf_override | Conf Override Settings | <QXZ>If populated, and the Template ID is set to --NONE-- then the contents of this field are used as the conf file entries for this phone. generate conf files for this phones server must be set to Y for this to work. This field should NOT contain the [extension] line, that will be automatically generated.</QXZ> |
| phones-group_alias_id | Group Alias ID | <QXZ>The ID of the group alias used by agents to dial out calls from the agent interface with different Caller IDs. no spaces or other special characters allowed. Must be between 2 and 20 characters in length.</QXZ> |
| phones-group_alias_name | Group Alias Name | <QXZ>The name used to describe a group alias, Must be between 2 and 50 characters in length.</QXZ> |
| phones-caller_id_number | Caller ID Number | <QXZ>The Caller ID number used in this Group Alias. Must be digits only.</QXZ> |
| phones-caller_id_name | Caller ID Name | <QXZ>The Caller ID name that can be sent out with this Group Alias. As far as we know this will only work in Canada on PRI circuits and using an IAX loop trunk through Asterisk.</QXZ> |
| servers-server_id | Server ID | <QXZ>This field is where you put the Asterisk servers name, does not have to be an official domain sub, just a nickname to identify the server to Admin users.</QXZ> |
| servers-server_description | Server Description | <QXZ>The field where you use a small phrase to describe the Asterisk server.</QXZ> |
| servers-server_ip | Server IP Address | <QXZ>The field where you put the Network IP address of the Asterisk server.</QXZ> |
| servers-active | Active | <QXZ>Set whether the Asterisk server is active or inactive.</QXZ> |
| servers-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> |
| servers-sysload | System Load | <QXZ>These two statistics show the loadavg of a system times 100 and the CPU usage percentage of the server and is updated every minute. The loadavg should on average be below 100 multiplied by the number of CPU cores your system has, for optimal performance. The CPU usage percentage should stay below 50 for optimal performance.</QXZ> |
| servers-channels_total | Live Channels | <QXZ>This field shows the current number of Asterisk channels that are live on the system right now. It is important to note that the number of Asterisk channels is usually much higher than the number of actual calls on a system. This field is updated once every minute.</QXZ> <QXZ>The Agents field shows the current number of agents logged into the agent screen on this server.</QXZ> |
| servers-disk_usage | Disk Usage | <QXZ>This field will show the disk usage for every partition on this server. This field is updated once every minute.</QXZ> |
| servers-system_uptime | System Uptime | <QXZ>This field will show the system uptime of this server. This field only updates if configured to do so by your administrator.</QXZ> |
| servers-asterisk_version | Asterisk Version | <QXZ>Set the version of Asterisk that you have installed on this server. Examples: 1.2, 1.0.8, 1.0.7, CVS_HEAD, REALLY OLD, etc... This is used because versions 1.0.8 and 1.0.9 have a different method of dealing with Local/ channels, a bug that has been fixed in CVS v1.0, and need to be treated differently when handling their Local/ channels. Also, current CVS_HEAD and the 1.2 release tree uses different manager and command output so it must be treated differently as well.</QXZ> |
| servers-max_trunks | Max Trunks | <QXZ>This field will determine the maximum number of lines that the auto-dialer will attempt to call on this server. If you want to dedicate two full PRI T1s to outbound dialing on a server then you would set this to 46. Any inbound or manual dial calls will be counted against this total as well. Default is 96.</QXZ> |
| servers-outbound_calls_per_second | Max Calls per Second | <QXZ>This setting determines the maximum number of calls that can be placed by the outbound auto-dialing script on this server per second. Must be from 1 to 100. Default is 5. Most SIP carriers can handle less than 4.</QXZ> |
| servers-telnet_host | Telnet Host | <QXZ>This is the address or name of the Asterisk server and is how the manager applications connect to it from where they are running. If they are running on the Asterisk server, then the default of localhost is fine.</QXZ> |
| servers-telnet_port | Telnet Port | <QXZ>This is the port of the Asterisk server Manager connection and is how the manager applications connect to it from where they are running. The default of 5038 is fine for a standard install.</QXZ> |
| servers-ASTmgrUSERNAME | Manager User | <QXZ>The username or login used to connect genericly to the Asterisk server manager. Default is cron</QXZ> |
| servers-ASTmgrSECRET | Manager Secret | <QXZ>The secret or password used to connect genericly to the Asterisk server manager. Default is 1234</QXZ> |
| servers-ASTmgrUSERNAMEupdate | Manager Update User | <QXZ>The username or login used to connect to the Asterisk server manager optimized for the Update scripts. Default is updatecron and assumes the same secret as the generic user.</QXZ> |
| servers-ASTmgrUSERNAMElisten | Manager Listen User | <QXZ>The username or login used to connect to the Asterisk server manager optimized for scripts that only listen for output. Default is listencron and assumes the same secret as the generic user.</QXZ> |
| servers-ASTmgrUSERNAMEsend | Manager Send User | <QXZ>The username or login used to connect to the Asterisk server manager optimized for scripts that only send Actions to the manager. Default is sendcron and assumes the same secret as the generic user.</QXZ> |
| servers-conf_secret | Conf File Secret | <QXZ>This is the secret, or password, for the server in the iax auto-generated conf file for this server on other servers. Limit is 20 characters alphanumeric dash and underscore accepted. Default is test. A strong conf file secret should be at least 8 characters in length and have lower case and upper case letters as well as at least one number.</QXZ> |
| servers-local_gmt | Server GMT offset | <QXZ>The difference in hours from GMT time not adjusted for Daylight-Savings-Time of the server. Default is -5</QXZ> |
| servers-voicemail_dump_exten | VMail Dump Exten | <QXZ>The extension prefix used on this server to send calls directly through agc to a specific voicemail box. Default is 85026666666666</QXZ> |
| servers-voicemail_dump_exten_no_inst | VMAIL Dump Exten NI | <QXZ>This is the dial plan prefix used to send calls directly to a user voicemail from a live call in the astGUIclient app. This is the No Instructions setting.</QXZ> |
| servers-answer_transfer_agent | auto dial extension | <QXZ>The default extension if none is present in the campaign to send calls to for auto dialing. Default is 8365</QXZ> |
| servers-routing_prefix | Routing Prefix | <QXZ>If populated, this value will be added in front of the auto dial extension when an auto-dial call is placced on a dialer server that is running Asterisk verison 13 or higher. Default is 13.</QXZ> |
| servers-conf_engine | Conferencing Engine | <QXZ>The conferencing engine used by Asterisk for agent sessions. Default is MEETME. CONFBRIDGE is only supported by patched Asterisk 16 and higher servers.</QXZ> |
| servers-conf_update_interval | Conf Update Interval | <QXZ>The interval of the conferences updating process. Default is 60 seconds.</QXZ> |
| servers-ext_context | Default Context | <QXZ>The default dial plan context used for scripts that operate for this server. Default is default</QXZ> |
| servers-sys_perf_log | System Performance | <QXZ>Setting this option to Y will enable logging of system performance stats for the server machine including system load, system processes and Asterisk channels in use. Default is N.</QXZ> |
| servers-vd_server_logs | Server Logs | <QXZ>Setting this option to Y will enable logging of all system related scripts to their text log files. Setting this to N will stop writing logs to files for these processes, also the screen logging of asterisk will be disabled if this is set to N when Asterisk is started. Default is Y.</QXZ> |
| servers-agi_output | AGI Output | <QXZ>Setting this option to NONE will disable output from all system related AGI scripts. Setting this to STDERR will send the AGI output to the Asterisk CLI. Setting this to FILE will send the output to a file in the logs directory. Setting this to BOTH will send output to both the Asterisk CLI and a log file. Default is FILE.</QXZ> |
| servers-balance_active | Balance Dialing | <QXZ>Setting this field to Y will allow the server to place balance calls for campaigns in the system so that the defined dial level can be met even if there are no agents logged into that campaign on this server. Default is N.</QXZ> |