help_documentation
Σύνολο εγγραφών στον πίνακα: 1581
| help_id | help_title | help_text |
|---|---|---|
| phones-on_hook_agent | On-Hook Agent Login | <QXZ>This option is only used for inbound calls going to an agent logged in with this phone. This feature will call the agent and will not send the customer to the agents session until the line is answered. Default is N for disabled.</QXZ> |
| phones-ASTmgrUSERNAME | Manager Login | <QXZ>This is the login that the GUI clients for this phone will use to access the Database where the server data resides.</QXZ> |
| phones-ASTmgrSECRET | Manager Secret | <QXZ>This is the password that the GUI clients for this phone will use to access the Database where the server data resides.</QXZ> |
| phones-login_user | Agent Default User | <QXZ>This is to place a default value in the agent user field whenever this phone user opens the client app. Leave blank for no user.</QXZ> |
| phones-login_pass | Agent Default Pass | <QXZ>This is to place a default value in the agent password field whenever this phone user opens the client app. Leave blank for no pass.</QXZ> |
| phones-login_campaign | Agent Default Campaign | <QXZ>This is to place a default value in the agent screen campaign field whenever this phone user opens the client app. Leave blank for no campaign.</QXZ> |
| phones-park_on_extension | Park Exten | <QXZ>This is the default Parking extension for the client apps. Verify that a different one works before you change this.</QXZ> |
| phones-conf_on_extension | Conf Exten | <QXZ>This is the default Conference park extension for the client apps. Verify that a different one works before you change this.</QXZ> |
| phones-park_on_filename | Agent Park File | <QXZ>This is the default agent screen park extension file name for the client apps. Verify that a different one works before you change this. limited to 10 characters.</QXZ> |
| phones-monitor_prefix | Monitor Prefix | <QXZ>This is the dial plan prefix for monitoring of Zap channels automatically within the astGUIclient app. Only change according to the extensions.conf ZapBarge extensions records.</QXZ> |
| phones-recording_exten | Recording Exten | <QXZ>This is the dial plan extension for the recording extension that is used to drop into meetme conferences to record them. It usually lasts upto one hour if not stopped. verify with extensions.conf file before changing.</QXZ> |
| phones-voicemail_exten | VMAIL Main Exten | <QXZ>This is the dial plan extension going to check your voicemail. verify with extensions.conf file before changing.</QXZ> |
| phones-voicemail_dump_exten | VMAIL Dump Exten | <QXZ>This is the dial plan prefix used to send calls directly to a user voicemail from a live call in the astGUIclient app. verify with extensions.conf file before changing.</QXZ> |
| phones-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> |
| phones-ext_context | Exten Context | <QXZ>This is the dial plan context that the agent screen, primarily uses. It is assumed that all numbers dialed by the client apps are using this context so it is a good idea to make sure this is the most wide context possible. verify with extensions.conf file before changing. default is default.</QXZ> |
| phones-phone_context | Phone Context | <QXZ>This is the dial plan context that this phone will use to dial out. If you are running a call center and you do not want your agents to be able to dial out outside of the agent screen application for example, then you would set this field to a dialplan context that does not exist, something like agent-nodial. default is default.</QXZ> |
| phones-codecs_list | Allowed Codecs | <QXZ>You can define a comma delimited list of codecs to be set as the default codecs for this phone. Options for codecs include ulaw, alaw, gsm, g729, speex, g722, g723, g726, ilbc, opus, slin, g719,... Some of these codecs might not be available on your system, like g729, g726 or opus. If the field is empty, then the system default codecs or the phone entry above this one will be used for the allowable codecs. Default is empty.</QXZ> |
| phones-codecs_with_template | Allowed Codecs With Template | <QXZ>Setting this option to 1 will include the codecs defined above even if a conf file template is used. Default is 0.</QXZ> |
| phones-conf_qualify | Conf Qualify | <QXZ>This setting allows you to add or remove the qualify entry in the Asterisk conf file for this phone if it is IAX type. Default is Y for active.</QXZ> |
| phones-dtmf_send_extension | DTMF send Channel | <QXZ>This is the channel string used to send DTMF sounds into meetme conferences from the client apps. Verify the exten and context with the extensions.conf file.</QXZ> |
| phones-call_out_number_group | Outbound Call Group | <QXZ>This is the channel group that outbound calls from this phone are placed out of. There are a couple routines in the client apps that use this. For Zap channels you want to use something like Zap/g2 , for IAX2 trunks you would want to use the full IAX prefix like IAX2/VICItest1:secret@10.10.10.15:4569. Verify the trunks with the extensions.conf file, it is usually what you have defined as the TRUNK global variable at the top of the file.</QXZ> |
| phones-client_browser | Browser Location | <QXZ>This is applicable to only UNIX/LINUX clients, the absolute path to Mozilla or Firefox browser on the machine. verify this by launching it manually.</QXZ> |
| phones-install_directory | Install Directory | <QXZ>Not used anymore.</QXZ> |
| phones-local_web_callerID_URL | CallerID URL | <QXZ>This is the web address of the page used to do custom callerID lookups. default testing address is: http://astguiclient.sf.net/test_callerid_output.php</QXZ> |
| phones-agent_web_URL | Agent Default URL | <QXZ>This is the web address of the page used to do custom agent Web Form queries. default testing address is defined in the database schema.</QXZ> |
| phones-nva_call_url | NVA Call URL | <QXZ>This is the optional web URL that can be used together with the NVA agi script in a Call Menu to log phone calls made outside of the agent screen. Variables that can be used with this feature are- phone_number, uniqueid, lead_id, extension, server_ip, entry_date, modify_date, status, user, vendor_lead_code, source_id, list_id, phone_number, title, first_name, middle_initial, last_name, address1, address2, address3, city, state, province, postal_code, country_code, gender, date_of_birth, alt_phone, email, security_phrase, comments, called_count, last_local_call_time, rank, owner, campaign_id, list_description, recording_id, recording_filename.</QXZ> |
| phones-nva_search_method | NVA Search Method | <QXZ>If this phone dials through the NVA agi script in a Call Menu, and the NVA agi option is set to use the phone NVA Search Method, this is where that is defined.</QXZ> |
| phones-nva_error_filename | NVA Error Filename | <QXZ>If this phone dials through the NVA agi script in a Call Menu, this is the error file that is played for the user of this phone if an error occurs.</QXZ> |
| phones-nva_new_list_id | NVA New List ID | <QXZ>If this phone dials through the NVA agi script in a Call Menu, this is the list ID that a new lead is inserted into if the phone number is not found and the NVA option to insert a new lead is set to Y. Default is 995.</QXZ> |
| phones-nva_new_phone_code | NVA New Phone Code | <QXZ>If this phone dials through the NVA agi script in a Call Menu, this is the phone code that a new lead is inserted with if the phone number is not found and the NVA option to insert a new lead is set to Y. Default is 1.</QXZ> |
| phones-nva_new_status | NVA New Status | <QXZ>If this phone dials through the NVA agi script in a Call Menu, this is the status that a new lead is inserted with if the phone number is not found and the NVA option to insert a new lead is set to Y. Default is NVAINS.</QXZ> |
| phones-AGI_call_logging_enabled | Call Logging | <QXZ>This is set to true if the call_log step is in place in the extensions.conf file for all outbound and hang up h extensions to log all calls. This should always be 1 because it is mandatory for many of the system features to work properly.</QXZ> |
| phones-user_switching_enabled | User Switching | <QXZ>Set to true to allow user to switch to another user account. NOTE: If user switches they can initiate recording on the new user phone conversation</QXZ> |
| phones-conferencing_enabled | Conferencing | <QXZ>Set to true to allow user to start conference calls with upto six external lines.</QXZ> |
| phones-admin_hangup_enabled | Admin Hang Up | <QXZ>Set to true to allow user to be able to hang up any line at will through astGUIclient. Good idea only to enable this for Admin users.</QXZ> |
| phones-admin_hijack_enabled | Admin Hijack | <QXZ>Set to true to allow user to be able to grab and redirect to their extension any line at will through astGUIclient. Good idea only to enable this for Admin users. But is very useful for Managers.</QXZ> |
| phones-admin_monitor_enabled | Admin Monitor | <QXZ>Set to true to allow user to be able to grab and redirect to their extension any line at will through astGUIclient. Good idea only to enable this for Admin users. But is very useful for Managers and as a training tool.</QXZ> |
| phones-call_parking_enabled | Call Park | <QXZ>Set to true to allow user to be able to park calls on astGUIclient hold to be picked up by any other astGUIclient user on the system. Calls stay on hold for upto a half hour then hang up. Usually enabled for all.</QXZ> |
| phones-updater_check_enabled | Updater Check | <QXZ>Set to true to display a popup warning that the updater time has not changed in 20 seconds. Useful for Admin users.</QXZ> |
| phones-AFLogging_enabled | AF Logging | <QXZ>Set to true to log many actions of astGUIclient usage to a text file on the user computer.</QXZ> |
| phones-QUEUE_ACTION_enabled | Queue Enabled | <QXZ>Set to true to have client apps use the Asterisk Central Queue system. Required for the system to work and recommended for all phones.</QXZ> |
| phones-CallerID_popup_enabled | CallerID Popup | <QXZ>Set to true to allow for numbers defined in the extensions.conf file to send CallerID popup screens to astGUIclient users.</QXZ> |
| phones-voicemail_button_enabled | VMail Button | <QXZ>Set to true to display the VOICEMAIL button and the messages count display on astGUIclient.</QXZ> |
| phones-enable_fast_refresh | Fast Refresh | <QXZ>Set to true to enable a new rate of refresh of call information for the astGUIclient. Default disabled rate is 1000 ms ,1 second. Can increase system load if you lower this number.</QXZ> |
| phones-fast_refresh_rate | Fast Refresh Rate | <QXZ>in milliseconds. Only used if Fast Refresh is enabled. Default disabled rate is 1000 ms ,1 second. Can increase system load if you lower this number.</QXZ> |
| phones-enable_persistant_mysql | Persistant MySQL | <QXZ>If enabled the astGUIclient connection will remain connected instead of connecting every second. Useful if you have a fast refresh rate set. It will increase the number of connections on your MySQL machine.</QXZ> |
| phones-auto_dial_next_number | Auto Dial Next Number | <QXZ>If enabled the agent screen will dial the next number on the list automatically upon disposition of a call unless they selected to PAUSE AGENT DIALING on the disposition screen.</QXZ> |
| phones-VDstop_rec_after_each_call | Stop Rec after each call | <QXZ>If enabled the agent screen will stop whatever recording is going on after each call has been dispositioned. Useful if you are doing a lot of recording or you are using a web form to trigger recording. </QXZ> |
| phones-enable_sipsak_messages | Enable SIPSAK Messages | <QXZ>If enabled, the server will send messages to the SIP phone to display on the phone display screen when logged into the agent web interface. Feature only works with SIP phones and requires sipsak application to be installed on the web server. Default is 0.</QXZ> |
| phones-DBX_server | DBX Server | <QXZ>The MySQL database server that this user should be connecting to.</QXZ> |