Blue.box to FreePBX - Blue

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Blue.box to FreePBX - Blue

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  • 6/1/13 Blue.box to FreePBX - blue.box - Confluence

    wiki2.2600hz.com/display/bluebox/Blue.box+to+FreePBX 1/2

    Blue.box to FreePBX

    The following example configures a SIP trunk between a FreeSWITCH server using Blue.Box and an Asterisk (v 1.6) server using

    FreePBX (v 2.8). Trunk registration will not be used. IP-based authentication will be used for connection with ACL routing controls.

    Blue.box Configuration

    SIP Interface Setup

    By default, Blue.Box creates two SIP interfaces during installation. For trunks that do not use username/password registration, the

    "Unauthenticated" SIP interface on port 5080 will be used. Ensure that the IP address information for the SIP interface is entered

    correctly to ensure external traffic is routed properly (see Configuring IP Addresses)

    The trunks (auto) network list (or ACL) is created by default during Blue.Box installation. Set the Inbound ACL for the Unauthenticated

    SIP Interface to trunks (auto). This will allow inbound calls from trunks on this interface to proceed through without challenging for a

    username and password.

    Lastly, specify a Default Incoming Context for the incoming calls from this SIP interface.

    Save changes to the Unauthenticated SIP Interface.

    Routing Setup

    NOTE: When setting up trunks with Blue.Box, creating routing patterns before creating trunks saves a lot of screen switching, but it

    can be done afterwards if so inclined.

    In the Simple Route module, a dial pattern for calls to route to the Asterisk trunk needs to be created. For the purposes of this guide,

    all 2XXX extensions are routed to the Asterisk server.

    After creating a new simple and entering a useful name for it, the following Short Hand dial pattern is entered: (2XXX). Create as many

    routes or add as many patterns as necessary for your routing scheme.

    Trunk/Gateway Setup

    In the Trunk Manager, create a new trunk for the Asterisk server. Provide the new trunk with a useful name for display and enter the IP

    address or hostname of the Asterisk server to be connected to in the Server field. The Trunk Type will be a SIP interface.

    In the Interface Management section, select the Unauthenticated SIP interface that was previously configured.

    To route outbound calls through the trunk to the Asterisk server, check off the patterns to be routed in the Route Outbound Calls

    Matching... section. If the required dial patterns are not yet available, ignore this and create them afterwards.

    Under Made From These Contexts..., specify which call contexts are allowed to be routed through this trunk. For internal use only,

    select "In-House Only" as the context to accept calls from.

    If desired, default outbound caller ID information can be specified for the trunk. This is not required and can be left blank.

    IMPORTANT: Make sure to lave the SIP Username, From Domain and SIP Password settings blank as registration will not be used

    and this may cause unexpected behavior.

    Save changes to the new trunk.

    FreePBX Configuration

    Trunk Setup

  • 6/1/13 Blue.box to FreePBX - blue.box - Confluence

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    In the Trunks section, create a new trunk for the FreeSWITCH server. Enter a useful display name for new trunk.

    Since FreePBX complains if a default outbound caller ID is not provided, enter a default caller ID to use for the trunk. This caller ID will

    only be used if the originating call does not provide any caller ID information.

    Since the trunk is not going to require registration on the PEER details in the Outgoing Settings needs to be entered. Use the following

    PEER details for the trunk:

    Routing Setup

    In the Outbound Routing section, create a new outbound route for calls to send to the FreeSWITCH trunk. Provide the route with a

    useful display name and enter in any required dial patterns for calls to send to the FreeSWITCH trunk. As an example, a dial pattern of

    1XXX will route all calls to extensions 1000 through 1999 to the provided destination.

    Lastly, provide the outbound route with a trunk sequence to try to send matched calls over. In this case, the trunk that was created for

    the FreeSWITCH server will be selected.

    Save changes to the new outbound route and reload Asterisk.

    Testing

    Test the dial patterns to the servers are passing calls off to each other.

    host=port=5080type=peerjbenable=nodisallow=allallow=ulawtrustrpid=yessendrpid=yescontext=from-trunkdeny=0.0.0.0/0.0.0.0permit=/255.255.255.255Save changes to the new trunk.