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The VoIP Providers feature is located in the Switchvox Administration portal under Setup -> Call Routing -> VoIP Providers. VoIP Providers may be used to connect to either a commercial VoIP phone service or other VoIP compatible hardware, such as Switchvox.
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You will first need to configure Switchvox to establish a connection to SIP provider. Click Create SIP Provider to add a new provider.
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In general, the only information that's required for a SIP provider is the provider information, which includes:
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From this page, you may edit or delete existing connections using the Action buttons on the right.
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These are the source ports used for establishing the RTP media pathway for SIP calls. It is unlikely that you will need to change this setting- only do so at your provider's request.
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In the Passthrough Outgoing Call Rule collector box, highlight the Internal rule, and click Allow. This allows extension to extension dialing. Note: if there is a need to share other outgoing rules, this is where you would allow access to those rules.
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In the Caller ID Settings tab, set Support Changing Caller ID to 'Yes'
Under the Connection Settings tab, set Qualify Hosts to 'Yes'. When this option is enabled, SIP option packets are sent every 60 seconds- this helps keep NAT tunnels open. In addition, latency responses are shown on the system status page.
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In the Call Settings tab, set G722 to 'ON' when using Digium phones (the remainder of the settings may be left to their default settings).
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