Call Routing Wizard

Call Routing Wizard Help

TheCall Routing Wizard is used to create a new pattern in the local call routing table. All pages of the call routing wizard provide the following buttons:

Previous leads back to the previous page.
Next leads forward to the next page. On the last page of the Call Routing wizard the button is changed into Finish to save all modifications.
Cancel leads back to the Call Routing table.
Help displays the online help.

Call Routing Wizard - Page 1 contains following components:

The Enable checkbox is used to enable the newly created routing rule. By default, this checkbox is selected, so the newly created routing rule will be enabled. But if you wish to create a routing rule for a later use, disable it from this page. The new routing rule will be added to the Call Routing Table but will be disabled and will not be considered when placing calls through the call routing unless it is enabled again.
The Destination Number Pattern text field specifies calls to which the rule should be applied. If a call, either inbound or outbound, has a destination number that matches the specified pattern, it will be completed according to the current rule. A routing pattern may contain wildcards. The complete list of characters and wildcards allowed in this text field is given on the Allowed Characters and Wildcards page.
The text field Number of Discarded Symbols requires the number of symbols that should be discarded from the beginning of the routing pattern. The field should be empty if no digits need to be discarded. Only digit values are allowed here otherwise the error message "Error: Number of Discarded Symbols is incorrect- digits allowed only." occurs.
Prefix requires entering the symbols (letters, digits and any characters supported in the SIP, H.323 username) that will be placed in front of the routing pattern instead of the discarded digits. The following tags can be used for this field:
<callerid:range> - used to apply the complete or a part of caller ID (the caller's number detected during the call) as a prefix. For example, <callerid:1-3> indicates that the first 3 digits of the caller ID will be considered as a prefix, <callerid:3-end> indicates that the caller ID from its 3rd digit and up to the end will be applied as a prefix. This tag can be used in combination with other digits at the beginning or at the end, as well as with wildcards.
<dialednum:range> - used to apply the complete or a part of dialed number (the number dialed by the caller to place a call) as a prefix. For example, <dialednum:1-3> indicates that the first 3 digits of the dialed number will be considered as a prefix, <dialednum:3-end> indicates that the dialed number from its 3rd digit and up to the end will be applied as a prefix. This tag can be used in combination with other digits at the beginning or at the end, as well as with wildcards.
The syntax aaa,,,bbb in the Prefix field allows for two-stage dialing. The aaa and bbb are the numbers to call; bbb can also be a series of digits to inject; a comma indicates a delay of one second. The syntax can be applied to include more call destination numbers separated by time intervals.
A two-stage dialing allows successive numbers to be dialed one after another with a delay in-between. For example, 11,,,11018 will call 11, wait until the call is established, wait for three seconds and then dial 11018. The capability of automatically dialing successive numbers allows the caller to bypass the IVR system on the call path and establish a direct call. The two-stage dialing is available for PBX, ISDN, and E1/T1 (if available on the model) destination types.
Suffix requires entering the symbols (letters, digits and any characters supported in the SIP username) that will be placed in the end of the routing pattern (ex: if the routing pattern is 12345, the Number of Discarded Symbols is 2, and the Prefix is 909 and the Suffix is 0a , the final phone number will be 9093450a).
Destination Type drop down gives you the option to select the destination type. The following destination types are available:
- PBX - local calls to QX200's extensions
- PBX-Voicemail - calls directly to the voice mailbox of the local PBX extension
- PBX-Intercom - local calls to PBX extensions with the request of Intercom service
- SIP - calls through a SIP server
- SIP_Trunk - calls through a SIP tunnels established (see SIP Tunnel Settings)
- IP-PSTN - calls through the IP-PSTN provider to the remove PSTN global telephone network
- FXO - calls to a PSTN global telephone network
- ISDN - calls to a ISDN global telephone network
The Metric text field provides a possibility to enter a rating for the selected route in a range from 0 to 20. If no value is inserted to this field, 10 will be taken as a default. In case two route entries match a user's dial string, the route with the lower metric will be preferred.
Description text field requires an optional description of the routing pattern.
The Filter on Source/Modify Caller ID checkbox selection allows to limit the functionality of the current route to be used by the defined caller(s) only. If this checkbox is enabled, source caller information (Source Number Pattern, Source Type, Source Port ID, etc.) will be required further in the Call Routing Wizard. This option is enabled by default.
The Set Date/Time Period(s) checkbox selection allows to define a validity period(s) for current routing pattern to take place and to define pattern date/time rules. When this checkbox is enabled, Call Routing Wizard - Date/Time Rules - Add Entry page will be displayed.
The Set Tracing/Debug Options on This Rule checkbox is used to switch events notification on the certain execution results of the corresponding routing rule. When this checkbox is enabled, Call Routing Wizard - Tracing/Debug Options page will be displayed.
The Set Overall Calling Time Limit checkbox selection allows to set and manage the overall duration of the calls placed with the selected routing rule.
Note:The Overall Calling Time Limitation checkbox is not allowed for PBX, PBX-Voicemail and PBX-Intercom destination types routing rules.
If this checkbox is not selected in the Call Routing Wizard first page, the overall call duration will be unlimited. When this checkbox is selected, Call Routing Wizard - Routing Overall Calls Limitation Settings page will be displayed.
Require Authorization for Enabling/Disabling checkbox is used to enable administrator's password authentication when enabler/disabler keys are configured for the routing rule. The service can be used locally from the handset (see Feature Codes) or remotely from Auto Attendant (see Auto Attendant Services). When this checkbox is selected, administrator's password will be requested to enable/disable the certain routing rule(s). If the administrator's password has been inserted incorrectly for 3 times, no status changes will be applied to any of the routing record(s), even to those which have no authorization enabled.
Enabler Key and Disabler Key text fields request digit combination which should be dialed from the handset or Auto Attendant to enable or disable the certain routing rules in the Call Routing Table. You can set the same Enabler/Disabler Key for multiple routing rules (the same key may be used as enabler for one routing rule, and as disabler for another one) - this will allow managing several routing rules with the single key.

Call Routing Wizard - Page 2 contains different components depending on the Destination Type selected on the previous step.

The Use Extension Settings drop down list is applicable to SIP and IP-PSTN destination types only and allows to select the extension (also Auto Attendant) on behalf of which the call will be placed. The SIP settings of the selected extension will be used as a caller information. If no entry is selected in this list, original caller information will be kept. When Keep original Caller ID checkbox is selected, called destination will receive the original caller's information, rather than the information of extension selected from the Use Extension Settings list.
When the checkbox Add Remote Party ID is selected, the Remote-Party-ID parameter is being delivered to the destination side upon call establishment procedure.
Destination Host text field indicates IP address or the host name of the destination (for direct call) or the SIP server (for calls through SIP server).
Destination Port text field requires port number of the destination or the SIP server.
User Name and Password - identification settings for public SIP servers or servers requiring authentication.
Enable Activity Timeout checkbox is used to limit time-to-live period of routing pattern (makes sense if accept or failure feedback arrives too late from the destination). Checkbox selection enables the Activity Timeout text field which is used to insert a routing pattern activity timeout (in the range from 1 to 180 seconds). When timeout is configured, the routing pattern will be active within the defined time frame and if no response has been received from the destination during that period, the pattern will be stopped and next routing rule might be optionally considered (depending on the Fail Reason configuration on the corresponding pattern).
Restrict the Number of Simultaneous Calls checkbox is only available for IP-PSTN destination type and is used to restrict the number of simultaneous calls to the public SIP server with the same username at the same time. This checkbox enables Allowed Call Count text field which requires the number of simultaneous calls allowed in a range from 1 to 64. If you leave this field empty, no limitation will apply to the number of simultaneous logons.
Use RTP Proxy checkbox is available for SIP and IP-PSTN dall types and is applicable only when route is used for calls through Quadro between peers both located outside the Quadro. When this checkbox Destination selected, RTP streams between external users will be routed through Quadro, otherwise, when checkbox is not selected, RTP packets will be moving directly between peers.
Voice Transcoding checkbox is available for SIP, H323 and IP-PSTN destination types and is meaningful when an extension is selected from the Use Extension Settings drop down list. Checkbox selection results the voice transcoding when performing IP calls through QX200. Independent on the codecs supported by the caller, QX200 will use the codecs of the extension selected from the Use Extension Settings drop down list to establish a call with the Destination. If the Destination supports either of the codecs configured for the used extension, the call between the caller and the Destination will be established by the first preferred codec of the Destination. If the Destination does not support either of the codecs configured for the used extension, the call will be disconnected.
The Collect Call checkbox is available only for E1/T1 and ISDN destination type and is used when it is simply preferable for the called phone to pay for the call. This service is applicabe only if the Collect Call checkbox is enabled on both calling and called party's IP PBXs.
The Single Call Duration Limit checkbox is available for SIP, IP-PSTN and PSTN destination types and is used to limit the duration of the call placed with the selected routing rule. If this checkbox is not selected, the call duration will be unlimited. This checkbox selection enables the Maximum Duration text field where the maximum duration of the call (in seconds) should be defined. Once the call duration reaches the value defined here, the call will be disconnected without prior notice.
A group of AAA Required checkboxes are used to choose one or more Authentication, Authorization, and Accounting (AAA) settings:
- Local Authentication - with this checkbox selected, callers will need to pass authentication through Local AAA Table when dialing the current pattern.
- RADIUS Authentication and Authorization - checkbox is present when RADIUS Client is enabled. With this checkbox selected, when dialing the current pattern, callers will need to pass the authentication through RADIUS Server.
- The RADIUS Accounting checkbox is accessible when the RADIUS Client is enabled. With this checkbox selected, no authentication will take place, but CDRs (call detail reports) of the calls made through this routing record will be sent to the RADIUS server. This checkbox selection enables the Client Code Identification checkbox.
- The Client Code Identification checkbox selection activates the code identification feature: a caller, after dialing the destination phone number, may optionally enter “*” and then an Identity Code. An Identity Code is an arbitrary digit string entered by the user to identify a specific call or call group. The Identity Code is sent with CDR to the RADIUS server and might be used by a billing program for grouping the calls having the same Identity Code.
If the authentication is configured based on the caller's address, callers will pass the authentication automatically; otherwise they will be required to identify themselves by a username and a password.
Attention: It is highly recommended to secure PSTN and IP-PSTN routing rules by selecting AAA Required options. Unsecured routing rules may cause unexpected expenses.
The Failover Reason(s) radio buttons indicate whether the system should use the next matching pattern if call setup with the current routing rule fails and allows choosing the reasons to be considered as a failover.
- None - indicates that matching patterns should not be used regardless of the failover reason.
- Failover Reason(s) - indicates possible failure reasons. Failure reasons vary depending on the destination type selected on the previous page. If the call cannot be established due to selected Failure Reasons, the call routing table will be parsed for the next matching pattern and, if found, the call will be routed to the specified destination.
  - Cannot Establish Connection - failure reason is available for FXO and ISDN calls only and indicates cases when connection cannot be established.
  - Busy - available for PBX, SIP, SIP Tunnel, and IP-PSTN destination types and indicates cases when the dialed destination is busy.
  - Wrong Number - available for PBX, SIP, SIP Tunnel, and IP-PSTN destination types and indicates cases when the dialed number is wrong.
  - Network Failure - available for SIP and IP-PSTN destination types and indicates cases when system overload, network failure or timeout expiration occurred.
  - System Failure - available for SIP, SIP Tunnel, and IP-PSTN destination types and indicates cases indicated in Network Failure and Other fail reasons.
  - Other - available for SIP, SIP Tunnel, and IP-PSTN destination types and indicates cases when authorization, negotiation, not supported or request rejected or other unknown errors occur.
- Any stands for all failure reasons mentioned in the Failover Reason(s) group.
The Custom Profile text field is present if the PBX-Voicemail destination type has been selected on the first page of the Call Routing Wizard. This field requires the Voice Mail Profile name to activate the custom voice mail settings on the extension when the corresponding routing rule will be used.
Note: If an extension does not have a profile specified here or the specified profile name is incorrect, the default Voice Mail Settings of the extension will be used.
Transport Protocol for SIP messages manipulation radio buttons group is available for SIP or IP-PSTN destination types only and allows to select the transport (UDP, TCP or TLS) to transmit the SIP messages through.
SIP Privacy manipulation radio buttons group is available for SIP destination type only and allows to select the security of the SIP route by means of hiding (or replacing, depending on the configuration of the SIP server) the key headers of the SIP messages used to establish the call.
- Default Privacy - with this selection, no QX200 specific SIP privacy will be applied, all privacy will be relied on the configuration of the SIP Server.
- Disable Privacy - with this selection, no SIP call security will be disabled, all headers of the SIP message will be transparently visible to the destination.
- Enable Privacy - with this selection, SIP privacy will be specified for the corresponding route. Selection enables a group of checkboxes to choose the key headers to be fully or partly hidden or replaced. Require Privacy checkbox selection is used to restrict the delivery of the SIP message if either of the selected headers cannot be hidden (or replaced, depending on the configuration of the SIP server) before being sent to the destination.

The Call Routing Wizard - Page 3 appears if Source Filter/Modify Caller ID Information checkbox is previously enabled in the Page 1 of the Call Routing Wizard and requires information about the source caller.

The Source Number Pattern field requires the caller address for which current route will be applied. The complete list of characters and wildcards allowed in this text field is given on the Allowed Characters and Wildcards page.
The Source Type drop down list gives a possibility to select the source type (PBX, SIP, H323, E1/T1) used by the source caller to reach the QX200.

The settings in Caller ID Modification group allow original Caller IDs of source callers to be modified.

The Number of Discarded Symbols text field requires the number of digits that should be discarded from the beginning of the Source Number Pattern. The field should be empty if no digits need to be discarded. Only digit values are allowed for this field, otherwise an error message occurs: "Error: Number of Discarded Symbols is incorrect - digits allowed only".

The Prefix text field requires entering the symbols (letters, digits and any characters supported in the SIP or H323 username) that will be placed in front of the Source Number Pattern instead of the discarded digits (ex: if the routing pattern is 12345, the Number of Discarded Symbols is 2, and prefix digits are 909, the final phone number will be 909345). Wildcard symbols are allowed here. The following tags can be used for this field:

<callerid:range> - used to apply the complete or a part of caller ID (the caller's number detected during the call) as a prefix. For example, <callerid:1-3> indicates that the first 3 digits of the caller ID will be considered as a prefix, <callerid:3-end> indicates that the caller ID from its 3rd digit and up to the end will be applied as a prefix. This tag can be used in combination with other digits at the beginning or at the end, as well as with wildcards.
<dialednum:range> - used to apply the complete or a part of dialed number (the number dialed by the caller to place a call) as a prefix. For example, <dialednum:1-3> indicates that the first 3 digits of the dialed number will be considered as a prefix, <dialednum:3-end> indicates that the dialed number from its 3rd digit and up to the end will be applied as a prefix. This tag can be used in combination with other digits at the beginning or at the end, as well as with wildcards.

Note: The Number of Discarded Symbols and Prefix text fields are hidden if E1/T1 destination type has been selected from the Call Routing Wizard - Page 1.

The Discard Non-Numeric Symbols checkbox is used to discard any non-numeric symbols from the Source Number Pattern.

The Display Name text field allows you to replace an original caller's ID with the custom display name for the corresponding routing rule. This field is optional and when it is left empty, an original caller ID will be displayed on the called destination's phone, otherwise the name inserted here will appear on the phone.

The Remove Display Name checkbox is used to remove original caller IDs from calls made with this routing rule.

Note: The Display Name and Remove Display Name fields are not available for PBX-Voicemail destination type routing rules.

The Next button will open a Call Routing Wizard - Page 4 where different information about source caller will be required depending on the selected Source Type. For the SIP Source Type, Source Host text field will require one or more IP addresses or host names of SIP server or H323 Gatekeeper where the caller is registered, or the caller's device in case of direct calls, separated by a space. In case of ISDN source type selected, Source Port ID drop down list will require to select the Trunk number, and on the next step, a list of timeslot(s) used to receive calls from the defined caller.

The Call Routing Wizard - Date/Time Rules - Add Entry page appears if the Set Date / Time Period(s) checkbox previously had been enabled on Page 1 of the Call Routing Wizard, and it will require information about the pattern validity period(s). Page provides selection between Typical and Custom date/time rule definition. Typical selection contains a group of radio buttons that are used to select the frequency of the corresponding routing pattern to take place:

Daily
Weekly - the preferred weekday(s) should be selected for this option.
Monthly - the calendar day should be selected for this option.
Annually - the calendar day and month should be selected for this option.

In Available Time Period drop down lists, the time range of the pattern validation should be defined. Any time selected in this field will be considered corresponding to the QX200's Time/Date Settings.

Custom selection provides a possibility to manually define the validity period(s). Use following format to insert pattern date/time rule(s): [Month,Month-Month,...][Day-Day,Day,...][hh:mm-hh:mm,...]; ...

The Call Routing Wizard - Routing Overall Calls Limitation Settings - Edit Entry page appears if the Set Overall Calling Time Limit checkbox previously had been enabled on Page 1 and allows to define the available duration of the calls with the selected routing rule as well as to specify the Expiration/Renewal Date for the available calls duration.
The Routing Overall Calls Limitation Settings - Edit Entry page consists of the following components:

The Available Calls Duration text field requires the maximum available duration of the calls(in minutes) placed with the selected routing rule. Once the Available Calls Duration reaches the value defined here, the current call will be disconnected without prior notice and no new call will be possible until this field is updated.
The Expiration/Renewal Date settings are used to configure the Expiration Date and Renewal Amount of the Available Calls Duration. Expiration/Renewal Date field provides selection between Periodic and Specific Date.
- The Periodic selection is used to define the expiration date of the allocated Available Calls Duration for the selected routing rule and has the following options:
- The Renewal Amount text field requires the renewal amount (in minutes) to be added to the Available Calls Duration when the expiration date of the Available Calls Duration is reached.
- The Discard remainder before renewal option selection allows to discard the remainder of Available Calls Duration before renewal and set the Renewal Amount as an available calls duration.
- The Specific Date selection provides a possibility to manually define the expiration date allocated for the Available Calls Duration for the selected routing rule. When the Specific Date expires, the selected routing rule becomes unavailable automatically and no new call will be possible until this field is updated.

The Call Routing Wizard - Tracing/Debug Options page appears if the Set Tracing / Debug Options on This Rule checkbox was previously enabled on Page 1 of the Local Call Routing Wizard. It will require information about the tracing/debug options. This page offers result options of the corresponding routing rule execution when the notification event will be printed in the Events page.

In Case of Successful Call - a notification event is printed when the successful call was established with the routing rule.
In Case of Failover - a notification event is printed when the call ends up on one of the failover reasons selected on the Page 2 of the Local Call Routing Wizard.
In Case if Call Failed to Establish - a notification event is printed when the call executed with the routing rule failed.

The Call Routing Wizard - Class of Services - Edit Entry page is used to assign the defined class of services to a certain call routing pattern. To use Class of Service feature for the corresponding routing rule, it should be enabled from the Class of Services page.

The Class of Service(CoS) functionality allows to permit or deny the attempt of extensions to use certain types of call routing rules.

Suppose you want for a certain group of PBX/Conference extensions to deny the right to make international calls, but allow them to make local and long distance calls and for another group of PBX/Conference extensions give a permission to make international calls only.

The classes defined in the Class of Services page will appear on this page to assign the corresponding routing rule to a certain class of service(s).

Note: The Class of Service feature is applicable only for PBX source type routing rules.

Note: The Filter on Source/Modify Caller ID option should be selected on the first page of the Call Routing Wizard to have a possibility to select the source caller type as a PBX.

Note: If the Class of Service is disabled, call routing rule(s) that are assigned to a certain CoS(s) will be available for any PBX extension, if there are no any other filtering limitations.