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Vegastream
The following example setup illustrates a working configuration of Vegastream 400 interoperating with SIPxchange; descriptions on the right provide details concerning entries made to the screen example on the left. Please refer to the Vegastream documentation for complete descriptions of each of the fields.
For sipX settings, access the sipX interface for the Configuration server, select Services from the left-hand main navigation menu, select Configuration, select Global Definitions from the screen that appears, and set the GW_DEFAULT_ADDR to the IP address of the Vegastream gateway in the associated field.
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<strong>LAN Configuration</strong>
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<li>Name: Vega400; the name of the gateway you want to use.</li>
<li>LAN Profile: 1</li>
<li>LAN Gateway: 10.1.20.1; the IP address of the gateway to get o other subnets.</li>
<li>Use DHCP Settings From Interface: 2; the interface you will use to get the DHCP information if you use DHCP.</li>
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<strong>LAN Profile 1</strong>
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<li>LAN Interface: The LAN interface you are using; in this case, 2.</li>
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<strong>LAN Interfaces</strong>
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<li>IP Address: This is the IP Address of the Vegastream. Set the static IP address here.</li>
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<strong>DNS Servers</strong>
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<li>Domain Name Server: The IP Address of the DNS server being used. Here the DNS server is being configured statically.</li>
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<strong>DSL Configuration</strong>
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<li>Network Type: The switch type you are using, such as ATT</li>
<li>Network Topology: T1 or E1</li>
<li>Line Encoding: B8ZS; must be B8ZS for ISDN. AMI or B8ZS for CAS.</li>
<li>Framing: ESF; must be ESF for ISDN, ESF or SF for CAS.</li>
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<strong>Port Configuration</strong>
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<li>You should not have to modify these fields.</li>
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<strong>Dial Planner (Profiles)</strong>
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<p>You will need at least two dial plans. One for calls coming in from the T1 interface, and the other for calls going out of the T1 interface. In this example, Profile IDs 2 ISDN_To_LAN and 3 LAN_to_ISDN_or_PBX are used (see the next Dial Planner profiles screens and associated field text).</p>
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<strong>Modify Plan</strong>
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<li>Name: The name of your plan (should have association to the original Dial Planner Profile name from the first graphic in this section).</li>
<li>Source: The source of the calls; in this example, all interfaces except the LAN.</li>
<li>Destination: The destination of the calls; in this example, the LAN.</li>
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<strong>Modify Plan</strong>
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<li>Name: The name of your plan (should have association to the original Dial Planner Profile name from the first graphic in this section).</li>
<li>Source: The source of the calls; in this example, all calls from the LAN.</li>
<li>Destination: The destination of the calls; in this example, destination T1 interface 2.</li>
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<strong>Media Capability</strong>
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<p>This is a list of the media capabilities supported by the Vegastream.</p>
<strong>G711 u-law Profiles</strong>
<ul>
<li>Packet Time: The sample size of the g711 sample.</li>
<li>VADU Enabled: VAD is enabled.</li>
<li>Out of Band DTMF: 1; this must be 1 for RFC 2833.</li>
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<strong>G729 AnnexA</strong>
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<li>Packet Time: The frame size of the g729a frame.</li>
<li>VADU Enabled: VAD disabled.</li>
<li>Out of Band DTMF: 1; this must be 1 for FRC 2833.</li>
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<strong>SIP Configuration</strong>
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<li>Default Proxy Hostname/IP: The host name of SIPxchange should be used.</li>
<li>Local Domain: The domain the Vegastream is in.</li>
<li>Local SIP Port: Port 5060 is the default.</li>
<li>Requrest URI Port: Port 5060 is the default.</li>
<li>LAN Profile: 1; this is profile 1, in this case.</li>
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<strong>Audio</strong>
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<p>You populate the Audio Profile fields with the coding algorithms RTP traffic is to use. Audio Profile 1 holds the first choice algorithm, Audio Profile 2 holds the second choice, etc.</p>
<li>Audio Profile 1: G711 Ulaw; this it he first coding algorithm.</li>
<li>Audio Profile 2: G729; this is the second coding algorithm.</li>
<li>Audio Profile 3: G711 Alaw; this is the third coding algorithm.</li>
<li>Audio Profile 4: None; no fourth choice is this example.</li>
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<strong>Miscellaneous</strong>
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<p>The defaults should remain intact. Check to make sure that your configuration is the same as shown here.</p>
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<strong>SIP Advanced</strong>
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<p>The defaults should remain intact. Make sure your configuration is the same as shown here.</p>
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