Switchvox VoIP Providers Admin Training

Owned by Merry Arpaia
Oct 22, 2024
3 min read

VoIP Providers

This article is a companion document for the Switchvox VoIP Providers video and will guide you through how to manage SIP providers, as well as RTP port range. 

 

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.

You will first need to configure Switchvox to establish a connection to SIP provider. Click Create SIP Provider to add a new provider.

In general, the only information that's required for a SIP provider is the provider information, which includes:

  • SIP Provider Name- create a name to refer to this provider (this is only used for referring to this provider in other areas of the admin suite).

  • Your Account ID- the account ID given to you by the provider. 

  • Your Password- the password for the account given to you by the provider. 

  • HostnameIP Address- the hostname, or the IP address for the SIP server, provided to you by your SIP provider. 

  • Callback Extension- the extension to ring when a voice call comes in over this provider if no incoming call rules are valid. Callback extensions may be any extension type (your reception phone, IVR, a queue, etc.).

  • Default Fax Extension- the extension to ring when a call comes in over this provider or channel and is identified as a fax. 

  • DTMF Mode- DTMF tones are the sounds emitted when you press buttons on your phone. 

 

Be sure to leave Host Type as 'Provider', unless instructed to change by your VoIP provider.

 

Determines your default caller ID for outgoing calls. Be sure your service providers accepts a modified caller ID, otherwise the outgoing call may fail. 

 

​​​​​​Set Qualify Hosts to 'Yes', allowing SIP option packets to be sent to the provider every 60 seconds- this helps keep NAT tunnels open. In addition, latency responses will be shown on the system status page. The remainder of the settings may be left to their default settings. Click Save SIP Provider when finished.

Note: You will be prompted to go back to the VoIP Providers menu or the Connection Status Page. If you choose Go To Connection Status, it will show if the SIP connection has been registered. If you select Back to VoIP Providers, you will be taken back to the VoIP Providers page. 

 

From this page, you may edit or delete existing connections using the Action buttons on the right.

 

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.

Click Create SIP Provider to create a SIP peer between two Switchvox systems. The information needed for a SIP Provider includes:

  • SIP Provider Name- create a name to refer to this peer. This will only be used when referring to this peer in other areas of the admin suite.

  • Your Account ID- create an account ID that will be used in both systems.

  • Your Password- create a password that will be used in both systems. 

  • Hostname IP Address- the hostname or the IP address of the far-end Switchvox.

  • Callback Extension- not used on the peer.

  • Default Fax Extension- not used on the peer.

 

Under the Peer Settings tab, set Host Type to 'Peer' and set Apply Incoming Call Rules to Provider to 'No'.

 

Under the Advanced Peer Settings, set Host is a Switchvox PBX and treat system's users like local users to 'Yes'.

 

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.

 

 

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. 

 

 

 

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).

 

 

Select Save SIP Provider when done. You will then be prompted to go back to the VoIP Providers menu.

 

 

Connection status will show if the SIP connection has been registered.

 

Unable to render {include} The included page could not be found.