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Here (Figure 4.34) is possible to configure which calls shall be routed to a SIP server which shall be responsible for routing them to their destiny. This routing is made through prefixes that may be inserted clicking the name of the route and then the link Insert above the prefixes table. To change or remove a route you only have to click its name and then the connection Modify or Delete, respectively.
To add a new SIP route click Insert and fulfil the following parameters:
Name: SIP server name;
Address: SIP server address;
Authentication: If it is necessary to make the authentication in the SIP server, you shall have to choose the option User/Password and fulfil the users name and respective password;
Proxy RTP4.18: It allows IPBrick to act as a proxy RTP and there is a NAT transposition. This option is automatically selected, if the route to be created is available for VoIP telephones in Internet;
Available to Internet: With this option selected, the route shall be available for VoIP telephones outside the LAN;
Symmetrical signalling: It allows to define if signalling is sent and received through the same door (port 5060);
Activate ENUM search: It allows IPBrick to search through ENUM.4.19
RTP proxy is a functionality supplied by IPBrick that allows to intermediate all the flow of RTP packets between tow VoIP terminals (or User agents in SIP terminology). It is used to transpose the NAT, i.e., when some VoIP terminal is "behind" a NAT. The prefixes inserted in this route shall be available automatically for the SIP telephones and the telephones connected to PBX. If there are additional interfaces and you intend to use a SIP route, it is necessary to add the route INTERFACE->INTERNET (for example PBX1->INTERNET or GSM->INTERNET), and include in that route a prefix matching the one of the route for the SIP server and include the prefix (in option Include prefix choose Yes).
Proxy RTP4.18