IMPORTANT
Use caution when making changes to settings on this page! You are urged to backup before making any changes!
Overview
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:
- From the PBX tab, select PBX Configuration
- From the left side column, select Advanced Settings
- Review the configured defaults
Review 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. Go to the Asterisk Dial Options section for details, 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. Go to the Asterisk Dial Options section for details, 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.
Making Changes
If you change a setting, you must save the change by clicking on the green check mark icon 
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.
Modules
Asterisk Builtin mini-HTTP server
| Field | Description |
|---|---|
| Enable Static Content | Default is False |
| Enable the mini-HTTP Server | Default is False. Set this to True to enable the mini-HTTP server |
| HTTP Bind Address | Default is blank. Enter a specific network interface for additional security |
| HTTP Bind Port | Default is 8088. Change this as required |
| HTTP Prefix | Default is blank |
Note that to begin using this interface you need to enable the AMI interface in the manager.conf file. Refer to Configuration File Editor for additional details.
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 built-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 |
|---|---|
| Call Forward Restrictions | This option is to determine which Call Restrictions are to be used when calling an extension that is forwarded to an external destination: caller or callee. Note: this option is used only for calls between extensions. |
| 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 minimum 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.
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. |
| SIP trunk inbound limit handling | This option sets the action(s) taken when the configured maximum SIP trunk channel limit (if any) is exceeded by a new inbound call. The options are:
|
Dialplan and Operational
NOTE
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. |
| Call Recording Policy | This policy is used to resolve the winner in a conflict between two extensions when one wants a call recorded and the other does not, if both their priorities are also the same. |
| 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 original incoming DID assuming there is a DID available. This is useful with some carriers that may require this under certain circumstances. |
| Voicecall Ignores DND | When this value is set to TRUE, intercom calls will be sent to extensions even when the extension has been set to DND (Do Not Disturb) by the user. This is a system-wide setting. Default value is FALSE. |
| 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 when not part of an intra-company route. See Asterisk Dial Options for details. |
| Country Indication Tones | Choose the country’s indication 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 dialed will be updated on the phone in the CNUM; so user sees the final manipulated number. |
| No Answer Timeout for Attended Transfer | The number of seconds to wait for an attended transfer destination to answer when using the Asterisk Attended Transfer function for a Blind Transfer. The default value is 15 seconds. |
| 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 processed 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. |
| Follow Me Restrictions | The default value for setting the dialing restrictions used to when deciding how to handle any outbound calls initiated by Find Me settings. Choices are to use the restrictions associated with the Caller (the party that originated the the call to the extension configured for Follow Me) or the Callee (the party that is configured for Follow Me). |
| Loop Prevention – Minimum DID Size | This option is used to configure the minimum size of DID numbers configured on the Inbound Routes page that should be checked in order to avoid Follow Me to DID loops. When this option is enabled and a configured DID is matched with a number in the Follow Me list, the number is skipped. The values are: disabled (default), 4, 5, 6, 7, 8, 9, 10, 11. |
Media Settings
NOTE
If you are using InfinityOne softphones and do not have ICE enabled, InfinityOne calls will ring, but no speech path will be established.
| Field | Description |
|---|---|
| ICE Support Enabled | Set this field to TRUE if you are configuring InfinityOne softphones on your system. (Does not require STUN to also be configured.) |
| STUN Server Hostname | A valid STUN server (such as stun.emetrotel.org:3478) may be required to be configured to resolve unusual network configurations that result in no speech path. Use as directed by E-MetroTel Support. |
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 specific applications. |
Time Condition Module
| Field | Description |
|---|---|
| Enable Maintenance Polling | If set to FALSE, this will override the execution of the Time Conditions maintenance task. |
| Maintenance Polling Interval | The polling interval in seconds used by the Time Conditions maintenance task. |
User Portal
| Field | Description |
|---|---|
| User Portal Enabled | Controls whether the UCX responds to https login requests to access the User Portal for any extension on the system (i.e. https://<ucxipaddress>/recordings). Set to True to enable the portal or False to disable it. |
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 configured with IMAP, so the IMAP username and password fields are displayed on the Extensions page. |
Asterisk Dial Options
The Asterisk Dial Options are defined in two fields:
- Asterisk Outbound Trunk Dial Options (for outgoing external calls)
- Asterisk Dial Options (for other types of calls)
The system wide settings for these options are defined in the Advanced Settings page under the Dialplan and Operational section. The values set should be appropriate for the majority of usage in the system to reduce the need to override them. The default system wide values on the UCx system are:
- Outbound Trunk Dial Options = Tt
- Asterisk Dial Options. = Ttr
The system wide values can be overridden in two places by checking the Override box beside the field:
- In the Trunk configuration page you can override the values for Asterisk Outbound Trunk Dial Options
- In the Extension configuration page you can override the values for Asterisk Dial Options
| Option | Description |
|---|---|
| D(called:calling) | Send the specified digits after the called party has answered, but before the call gets bridged. The ‘called’ digits are sent to the called party, and the ‘calling’ digits are sent to the calling party. Both arguments can be used alone. |
| h | Allow the called party to hang up by using the In-Call Asterisk Disconnect code (default value is **) |
| H | Allow the calling party to hang up by using the In-Call Asterisk Disconnect code (default value is **) |
| i | Any forwarding requests that may be received on this dial attempt will be ignored. |
| I | Any connected line update requests or any redirecting party update requests that may be received on this dial attempt will be ignored. |
| r | Generate ringing to the calling party, even if the called party is not actually ringing. Pass no audio to the calling party until the called channel has answered. |
| S(x) | Hang up the call x seconds after the called party has answered the call. |
| t | Allow the called party to transfer the calling party by using the In-Call Asterisk Blind Transfer code (default value is ##) |
| T | Allow the calling party to transfer the called party by using the In-Call Asterisk Blind Transfer code (default value is ##) |
| w | Allow the called party to enable recording of the call by using the In-Call Asterisk Toggle Call Recording code (default value is *1) |
| W | Allow the calling party to enable recording of the call by using the In-Call Asterisk Toggle Call Recording code (default value is *1) |
NOTE
Checking the Override box clears the default system wide value and makes the field editable. Make sure the desired options are entered in the field. If left blank, call features like Call Park that require the Blind Transfer capability will not work.