Advanced Settings (6.0)

The Advanced Settings page contains settings that are applied to the entire UCx system. One of the first steps in the configuration of your UCx system is to review the Advanced PBX settings.

To view the Advanced settings of your UCx system, perform the following steps:

  1. Open the UCx Web-based Configuration Utility
  2. From the PBX tab, select PBX Configuration
  3. From the left side column, select Advanced Settings
  4. Review the configured defaults
IMPORTANT: Use caution when making changes to these settings! You are urged to backup before making any changes!

If you change a setting, you must save the change by clicking on the green check box that appears on the right hand side of the setting. 

You can restore the default value by clicking on the green arrow icon. This icon is displayed if the value is not the default. 

The example below shows the green icons after a change is made to one of the settings.

You should review and configure at least the following settings:

  • Asterisk Dial Options - Options to be passed to the Asterisk Dial Command when making internal calls or for calls ringing internal phones. The options are documented in the Asterisk Dial Options page, a subset of which are described here. The default options T and t allow the calling and called users to transfer a call with ##. The r option allows Asterisk to generate ringing back to the calling phones which is needed by some phones and sometimes needed in complex dialplan features that may otherwise result in silence to the caller
  • Asterisk Outbound Trunk Dial Options - Options to be passed to the Asterisk Dial Command when making outbound calls on your trunks when not part of an Intra-Company Route. The options are documented in Asterisk Dial Options, a subset of which are described here. The default options T and t allow the calling and called users to transfer a call with ##. It is HIGHLY DISCOURAGED to use the r option here as this will prevent early media from being delivered from the PSTN and can result in the inability to interact with some external IVRs
  • Country Indications Tones - select the localized indications for your country/region

 

The Advanced Settings panel contains settings for the following modules:

Call Flow Control Module

Field Description
Hook Time Conditions Module By default, the Call Flow Control module will not hook Time Conditions allowing one to associate a call flow toggle feature code with a time condition since time conditions have their own feature code. If there is already an association configured (on an upgraded system), this will have no affect for the Time Conditions that are effected. Setting this to TRUE reverts the behaviour by allowing the use of a call flow toggle to be associated with a time condition.
This can be useful for two scenarios:
First, to override a Time Condition without the automatic resetting that occurs with the buil-in Time Condition overrides.
Second is the ability to associate a single call flow toggle with multiple time conditions thus creating a master switch that can be used to override several possible call flows through different time conditions.

Call Forward Module

Field Description
Create Call Forward Hints Setting this flag will generate the required dialplan that allows users to subscribe to BLF notifications about the device call forwarding status.

ChanSpy Direct Settings

Field Description
ChanSpy Direct Excluded Extensions A list of extensions to be excluded from the ChanSpy Direct Feature (i.e. extensions that cannot be listen to). enter a comma separated list of extensions and/or extension ranges (do not use spaces).
For example: 221,222,240-249,299
ChanSpy Direct Password Password required to listen to or listen & whisper to a specific extension.

DAHDI Configuration Module

Field Description
Allow PRI Discrete Channels DAHDI trunk configuration is normally done using groups for PRI configuration. if there is a need to configure trunks to specific channels, setting this to TRUE will allow each channel to be configured. This can be useful when troubleshooting a PRI and trying to isolate a bad B Channel.
Software EC Software EC to be used in system.conf.

Device Settings

Field Description
Require Strong Secrets Requires a strong secret on SIP and IAX devices with at least two numeric and non-numeric characters, and a mininum of 6 characters. 
SIP canreinvite (directmedia) Default setting for SIP canreinvite (same as directmedia). See Asterisk documentation for details.
SIP trustrpid Default setting for SIP trustrpid. See Asterisk documentation for details.
SIP sendrpid Default setting for SIP sendrpid. A value of YES is equivalent to RPID and will send the Remote-Party-ID header. A value of PAI will send the P-Asserted-Identity header. See Asterisk documentation for details.
SIP nat

Default setting for SIP nat.

  • YES will attempt to handle nat, also works for local (uses the network ports and address instead of the reported ports).
  • NO follows the protocol.
  • NEVER tries to block it, no RFC3581.
  • ROUTE ignoes the rport information. 

See Asterisk documentation for details.

SIP encryption Default setting for SIP encryption. Whether to offer SRTP encrypted media (and only SRTP encrypted media) on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if the peer does not support SRTP. See Asterisk documentation for details.
SIP qualifyfreq Default setting for SIP qualityfreq. Frequecy that 'quality' OPTIONS messages will be sent to the device. Can help to keep NAT holes open but not dependable for remote client firewalls. See Asterisk documentation for details. 
SIP and IAX qualify Default setting for SIP and IAX qualify. Whether to send periodic OPTIONS messages (for SIP) or otherwise monitor the channel, and at what point to consider the channel unavailable. A value of YES is equivalent to 2000, time in msec. Can help to keep NAT holes open but not dependable for remote client firewalls. See Asterisk documentation for details. 

Dialplan and Operational

Use caution when changing the setting for Disallow Transfers for Inbound Callers. Read the notice Security exploit of the Transfer functionality (FREEPBX-12058) before making the change.

Field Description
Block CNAM on External Trunks Some carriers will reject a call if a CallerID Name (CNAM) is presented. This occurs in several areas when configuring CID in the PBX using the format of 'CNAM'. To remove the CNAM part of CID on all external trunks, set this value to TRUE. This WILL NOT remove CNAM when a trunk is called from an intra-company route. This can be done on each individual trunk in addition to globally if there are trunks where it is desirable to keep CNAM information, though most carriers ignore CNAM.
Generate Diversion Headers If this value is set to TRUE, then calls going out on outbound routes that originate from outside the PBX and were subsequently forwarded through a call forward, ring group, follow-me or other means, will have a SIP diversion header added to the call with the orginal incoming DID assuming there is a DID available. This is useful with some carriers that may require this under certain circumstances.
Asterisk dial Options Options to be passed to the Asterisk Dial command when making internal calls or to calls ringing internal phones. See Asterisk Dial Options for details.
Asterisk Outbound Trunk Dial Options Options to be passed to the Asterisk Dial command when making outbound calls on trunks whne not part of an intra-company route. See Asterisk Dial Options for details.
Country Indication Tones Choose the country's indicaation tones to use when creating the different standard telephony tones such as ringing, busy, congestion, etc.
Disallow Transfers for Inbound Callers Disallow transfer features (by default ## and *2) for callers who pass through inbound routes (e.g. external callers).
Display CallerID on Calling Phone When set to TRUE and when CONNECTEDLINE() capabilities are configured and supported by the phone, the CID value being transmitted on this call will be updated on the phone in the CNAM field prepended with CID; so user knows what is being presented to the caller if the outbound trunk supports setting the transmitted CID.
Display Dialed Number on Calling Phone When set to TRUE and when CONNECTEDLINE() capabilities are configured and supported by the phone, the number actually dialled will be updated on the phone in the CNUM; so user sees the final manipulated number.
Ringtime Default Default number of seconds to ring phones before sending callers to voicemail or other destinations. This can be set per extension. Extensions with no voicemail or other destinations will ring indefinitely.
Speaking Clock Time Format Time format to use with the Speaking Clock.
Digital Trunk Inbound Delay A processing delay after receiving an inbound call over a digital trunk.
Enable this delay if the digital trunk (PRI) is connected to a CO that sends the caller ID number (CNUM) in the setup message and the caller ID name (CNAM) in a following facility message. When such a call is routed to a SIP endpoint (trunk or phone) without a delay, the caller ID name is not presented to the SIP endpoint until the call is answered. Configuring a short delay allows the facility message to be procesed before routing the call to the SIP endpoint - the caller ID name is then presented when the call starts ringing at the SIP endpoint.

Follow Me Module

Field Description
Create Follow Me at Extension Creation Time When creating a new extension, setting this to TRUE will automatically create a new Follow-Me for that extension using the default settings listed below.
Disable Follow Me Upon Creation Default value for Follow-Me Disable setting. When first creating a Follow-Me or if auto-created with a new extension, setting this to TRUE will disable the Follow-Me setting which can be changed by the user or admin later.
Default Follow Me Ring Time The default Follow-Me Ring Time when first created or if auto-created with a new extension.
Default Follow Me Initial Ring Time The default Follow-Me Initial Ring Time when first created or if auto-created with a new extension.
Default Follow Me Ring Strategy The default Follow-Me Ring Strategy when first created or if auto-created with a new extension.

Media Settings

Field Description
ICE Support Enabled Set this field to TRUE if you are configuring UCx WebRTC clients on your system.
STUN Server Hostname If ICE support is enabled, you must configure a STUN Server for ICE to be operational. Enter the hostname of the STUN Server to be used by the ICE feature.

Queues Module

Field Description
Hide Queue No Answer Option

The field Queue No Answer is hidden by default. To display the field, you have to change this setting to False.

Agent Called Events Default Default state for AMI events related to an agent's call. This setting only affects the default for NEW queues, does not change existing queues.
Member Status Event Default Default state for AMI QueueMemberStatus event. This setting only affects the default for NEW queues, does not change existing queues.
Stop Recording After Transfer When an external incoming queue call that is being recorded is transferred by an agent, you can choose either to stop recording the call or to continue recording the call.

System Setup

Field Description
Call Recording Format Format to save recorded calls for most call recording unless specified differently in specfiic applications.

Time Condition Module

Field Description
Enable Maintenance Polling If set to FALSE, this will override the execution of the Time Conditions maintenance task.
Maintanence Polling Interval The polling interval in seconds used by the Time Conditions maintenance task.

Voicemail Module

Field
Description
Mask Voicemail Passwords This field controls the display of the voicemail password, "masked" or "unmasked" in the Extensions page. By default the password is unmasked (False).
Create Voicemail Hints Set this to TRUE to allow the programming of busy lamp fields (BLF) for voicemail boxes on your phone.
Provide IMAP Voicemail Fields Set this to TRUE if you have voicemail configued with IMAP, so the IMAP username and password fields are displayed on the Extensions page.

Self Assessment Quiz

  1. If the Asterisk Outbound Trunk Dial Options contains the parameter tT, what does dialing ## do when a phone is engaged in a call?
  2. Do you know the implications of changing the setting for Disallow Transfers for Inbound Callers?
  3. Have you reviewed the Asterisk Dial Options, do you know what the wW option is for?
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Release 6.0
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