Data sheet
Table Of Contents
- DOCUMENT PURPOSE
- CHANGE LOG
- Firmware Version 1.0.19.27
- Firmware Version 1.0.19.21
- Firmware Version 1.0.18.13
- Firmware Version 1.0.18.12
- Firmware Version 1.0.18.9
- Firmware Version 1.0.17.16
- Firmware Version 1.0.16.20
- Firmware Version 1.0.16.18
- Firmware Version 1.0.15.16
- Firmware Version 1.0.14.24
- Firmware Version 1.0.14.23
- Firmware Version 1.0.14.21
- Firmware Version 1.0.13.14
- Firmware Version 1.0.12.19
- Firmware Version 1.0.11.27
- Firmware Version 1.0.0.7
- WELCOME
- PRODUCT OVERVIEW
- INSTALLATION
- GETTING STARTED
- SYSTEM SETTINGS
- PROVISIONING
- EXTENSIONS
- EXTENSION GROUPS
- ANALOG TRUNKS
- VOIP TRUNKS
- SLA STATION
- CALL ROUTES
- CONFERENCE
- VIDEO CONFERENCE
- IVR
- LANGUAGE SETTINGS FOR VOICE PROMPT
- VOICEMAIL
- RING GROUP
- PAGING AND INTERCOM GROUP
- CALL QUEUE
- PICKUP GROUPS
- MUSIC ON HOLD
- FAX SERVER
- BUSY CAMP-ON
- PRESENCE
- FOLLOW ME
- SPEED DIAL
- DISA
- EMERGENCY
- CALLBACK
- BLF AND EVENT LIST
- DIAL BY NAME
- ACTIVE CALLS AND MONITOR
- CALL FEATURES
- PBX SETTINGS
- SIP SETTINGS
- IAX SETTINGS
- INTERFACE SETTINGS
- API Configuration
- CTI SERVER
- ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)
- CRM INTEGRATION
- PMS INTEGRATION
- WAKEUP SERVICE
- ANNOUNCEMENTS CENTER
- STATUS AND REPORTING
- USER PORTAL
- MAINTENANCE
- EXPERIENCING THE UCM6200 SERIES IP PBX
P a g e | 146
UCM6200 Series User Manual
Version 1.0.19.27
Skip Voicemail
Password
Verification
When user dials voicemail code, the password verification IVR is skipped. If enabled,
this would allow one-button voicemail access. By default, this option is disabled.
Music On Hold
Select which Music On Hold class to suggest to extensions when putting them on
hold.
Enable LDAP
If enabled, the batch added extensions will be added to LDAP Phonebook PBX list; if
disabled, the batch added extensions will be skipped when creating LDAP
Phonebook.
Enable WebRTC
Support
If enabled, extensions will be able to login to user portal and use Web RTC features.
Call Duration Limit
Configure the maximum duration of call-blocking.
SIP Settings
NAT
Use NAT when the PBX is on a public IP communicating with devices hidden behind
NAT (e.g., broadband router). If there is one-way audio issue, usually it's related to
NAT configuration or Firewall's support of SIP and RTP ports.
The default setting is enabled.
Can Direct Media
By default, the PBX will route the media steams from SIP endpoints through itself. If
enabled, the PBX will attempt to negotiate with the endpoints to route the media
stream directly. It is not always possible for the PBX to negotiate endpoint-to-endpoint
media routing. The default setting is "No".
DTMF Mode
Select DTMF mode for the user to send DTMF. The default setting is "RFC2833". If
"Info" is selected, SIP INFO message will be used. If "Inband" is selected, 64-kbit
codec PCMU and PCMA are required. When "Auto" is selected, RFC2833 will be
used if offered, otherwise "Inband" will be used.
Enable Keep-alive
If enabled, empty SDP packet will be sent to the SIP server periodically to keep the
NAT port open. The default setting is "Yes".
Keep-alive
Frequency
Configure the number of seconds for the host to be up for Keep-alive. The default
setting is 60 seconds.
TEL URI
If the end device/phone has an assigned PSTN telephone number, this field should
be set to "User=Phone". Then a "User=Phone" parameter will be attached to the
Request-Line and TO header in the SIP request to indicate the E.164 number. If set
to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting
is disabled.