General options

The following fields in Options (Figure ):

Figure 5.40: VoIP - Options

• Router with full DNAT?: If IPBrick is connected to a router responsible for the access to the exterior (in terms of VoIP) that allows the 'passage' of all traffic, it is necessary to select Yes and indicate the external address of that same router in Router public IP address;
• IP address of the IPBrick public interface used by the VoIP service: IP address of the public interface of IPBrick responsible for the VoIP service;
• Intranet VoIP Server only?: It allows to route the network traffic only in a interface and not in two interfaces, as usual;
• Remove default national prefix (0): It removes national prefix normally used;
• Get call source address from IPBrick LDAP: If activated, it goes to the database LDAP of IPBrick defined in IPBrick IP address and in IPBrick DNS domain and, if it finds the calling number in the database, it will replace it by the name of the entity associated to that number.
• Immediate answer on calls originated in a PBX: It is advisable to have this option connected if you are using connections to SIP servers (ex: VoIPBuster, NetCall), in order to avoid timeouts in the PBX central. If, for example, you intend to define rates for the calls from the PBX, this option shall have to be deactivated to avoid that the user starts paying as soon as he dials the number.
• Attendance Timeout: Time (seconds) during which the call is sent to the destiny phone, before being sent or routed to another phone;
• Call Timeout: Time (seconds) during which the connection is trying to be established. If it expires, the attempt will be ended;
• Timeout to hangup calls without sound;
• Timeout to hangup calls on hold without sound;
• Enable SIP video support: Enables the support for SIP video;
• Attended transfer: If yes you can define a key activation sequence to do a attended transfer. So you can stop using this feature from the SIP phone and use it from the VoIP server;
• Blind transfer: If yes you can define a key activation sequence to do a blind transfer. So you can stop using this feature from the SIP phone and use it from the VoIP server;
• Voicemail: Enables general voicemail for VoIP;
• Call's prioritization: If enabled it will be possible to define priority levels for each route prefix defined in Routes Management. The level is from 1 (highest) to 10 (lowest). Example: In a LAN-PSTN route all the BRI lines are full. If a emergency call prefix (911) have maximum priority defined, when someone dial 911 some current call can be disconnected;
• Store calls details records in csv file: All the call history in the default asterisk format will be saved to a file called Master.csv. This file can be downloaded acceding by ftp with username voipcdr and password L1opardo;
• IP of server-signalling different from the media server: If a remote signalling service is running in one server, and the remote media server is running in a different one, this option must be activated;