Installation guide
QuadroM32x/8L/26x/12Li/26xi Manual II: Administrator's Guide  Administrator’s Menus 
QuadroM32x/8L/26x/12Li/26xi; (SW Version 5.3.x)       
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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. 
The Play audible signal before Intercom activation checkbox is appeared only if PBX Intercom is selected as Destination Type (see Manual III 
– Extension User’s Guide-Intercom Service). 
The AAA Required checkboxes are used to choose one or more of the following Authentication, Authorization, and Accounting (AAA) settings: 
•  Local Authentication –  with this checkbox selected, callers will need to pass authentication through the Local AAA Table when dialing the 
current pattern. 
•  RADIUS Authentication and Authorization – this checkbox is present when a RADIUS client is enabled. With this checkbox selected, callers 
will need to pass the authentication through RADIUS server (see above) when dialing the current pattern. 
•  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. 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. 
•  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.
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 Check with 3PCC checkbox is used to request a 3PCC approval before placing a call with the specific routing rule. When this checkbox is 
selected and the corresponding routing rule is used to place a call, Quadro sends a request to the call controlling application for the managing 
person to accept or reject the specific call (it can be a popup window or any other type of dialog box, depending on the call controlling application). If 
the request is accepted, the call will be placed. Otherwise, if the request is rejected, the call will be skipped. In case of no feedback from the call 
controlling application, the call will be accepted after a timeout defined in the configuration of the call controlling application. 
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. 
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, SIP Tunnel, 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. 
Cannot Establish Connection – available for FXO, ISDN and E1/T1 destination types and indicates cases when connection cannot be 
established. 
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 (see 
Voice Mail Profiles) on the extension when the 
corresponding routing rule will be used.  
Please 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. 
The Transport Protocol for SIP messages manipulation radio buttons group is available for SIP or IP-PSTN destination types only and allows you 
to select the transport (UDP, TCP or TLS) to transmit the SIP messages through.  
The SIP Privacy manipulation radio buttons group is only available for the SIP destination type and allows you 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, Quadro specific SIP privacy will not be applied and all privacy will rely on the configuration of the SIP 
Server. 










