VOIP Providers

This tool lets you manage your Voice Over Internet Protocol (VOIP) Providers. VOIP providers can be either commercial VOIP phone services or other VOIP hardware such as a peered Switchvox.

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• 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. Enter the account ID provided to you by your SIP provider. This may also be called Login, or Authentication ID.

• Your Password. Enter the password for the account.

• Hostname/IP Address. Enter the hostname or IP address for the SIP server provided to you by your SIP provider. This may also be called Registrar Server.

• Callback Extension. Enter the extension to ring when a voice call comes in over this provider or channel. Callback extensions can be any extension type (your receptionist’s phone, an IVR, etc.).

• Default Fax Extension. Enter the extension to ring when a call comes in over this provider or channel and is identified as a fax. (Switchvox only identifies T.38 faxes over SIP providers. A fax coming in over an IAX provider or a channel, or a fax using a different standard may not be identified as a fax.) You must install a fax license and the fax software to make faxing available. (For details, see the “Digium Addon Products” article.)

• DTMF Mode. DTMF tones are the sounds emitted when you press buttons on your phone. Options in the dropdown include: RFC4733, Info, and Inband.

  • SIP extensions, refer to your phone's user manual for the DTMF mode that your phone uses.

  • SIP providers, ask your provider which DTMF mode it supports.

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• Supports Changing Caller ID. Changing your caller ID information may cause all outbound calls to fail, so be careful when using this setting. Be sure that your provider accepts a modified Caller ID, then enable this option and customize the Caller ID displayed on outbound calls.

• Caller-ID method. Leave this set to From Header unless your provider explicitly tells you to use a different setting. The vast majority of providers require caller ID to be supplied in the From header.

• Caller ID Name / Caller ID Number. Enter a name and phone number to display as your Caller ID when making outbound calls with this provider.

Connection Settings Tab

The Connection settings determine how the connection is established with this provider.

• SIP Port. This is the UDP port used to communicate with this SIP Provider. By default this is set to 5060, the standard port for SIP communication. This port can bechanged if your provider or peer requires it.

• SIP Expiry. This controls the duration (in seconds) between SIP registration requests sent to your provider. Selecting a value other than 120 may cause problems with NAT routers. Do not change this unless explicitly instructed by your provider.

• Proxy Host. If your SIP provider says you need to use a SIP proxy, enter the proxy host IP here.

• Authentication User. If your SIP provider says you need a different authentication name than your account name, enter that authentication name here.

• Always Trust this Provider. Most providers require this option be set to Yes. It is a little less secure because it does not require authentication on incoming calls from this provider. If you set this to No and it does not work properly, set it back to Yes.

• Qualify Hosts. Enabling this option sends a SIP Options packet to this SIP provider every 60 seconds. This keeps NAT tunnels open. Response latency for these requests are shown in the System Status page. Note: Do not enable this option if your provider does not respond to SIP Options packets.

• Include user=phone in SIP. This option has no effect on most SIP providers. Enable it if your provider explicitly requests it. This adds the string “user=phone” on the end of the SIP URI used in various messages.

• Use Local Address in From Header. Leave this set to No. Changing it to Yes will

• violate section 8.1.1.3 of the SIP standard, RFC 3261. Only set this to Yes if your provider explicitly requests that you violate the SIP standard in this manner. If this is Yes, Switchvox uses its own local IP address in the From header of requests.

• SIP Provider Host List. Some SIP providers have multiple servers on the internet that communicate with Switchvox. For example, you may register sip.example.com but receive calls from ny.example.com, la.example.com, and mia.example.com. Use this option to list multiple hosts (or IP addresses) so that Switchvox can route calls from them. If you do not know if there are multiple hosts, ask your provider. If there are multiple hosts, ask your provider to supply you with a list of their IP addresses.

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The following table describes the audio codecs. If you entered a license key for the G729 Codec inTools > Digium Addon Products, you will also see G729 in the list of codecs.
 

• Video Codecs. The video codecs H.263, H.263+, and H.264 are available.

• Map Distinctive Rings. Some providers let users set different ring tones based on the dialed number, but do not pass the dialed number information directly. You may have to experiment with this setting and the settings of your SIP provider to get the right phone number associated with the right ring type.

• Call Settings.

  • Enable Jitterbuffer. Enable this option to combat network problems. A jitterbuffer induces a small delay in call latency to make up for lost or delayed voice packets. You may want to enable the jitterbuffer if you have bad network connectivity between Switchvox and the Provider. The options are as follows:

    • Never: It’s best to leave the jitterbuffer disabled if you are not having any audio quality problems.

    • When Needed: Intelligently enable the jitterbuffer with a dynamic window size when the call is terminated on Switchvox. Try this option first if you have audio quality issues.

    • Always: Always enable the jitterbuffer with a dynamic window size. This may be useful if you are using phones or other SIP devices that do not have their own jitterbuffer.

    • When Needed with Fixed Window: Intelligently enable the jitterbuffer with a fixed window size. This may be useful if jitter is sporadic, yet sudden. Or, if you need to mask problems with poorly configured routers that cannot implement QoS.

    • Always with Fixed Window: Always enable the jitterbuffer with a fixed window size. Select this if all else fails. Note that this just masks over the underlying issue. It’s best to fix the jitter problem itself, instead of using this option.

  • Allow Reunite. We strongly recommend leaving this set to Never. Enabling this option is risky and will likely result in sporadic one-way audio, dropped calls, or not being able to place calls. This lets Switchvox redirect the media stream of some calls, bypassing the PBX and passing directly between both participants. This may offer bandwidth and quality improvements in certain scenarios, but it is extremely risky.

    • Never: This is the safest setting that presents the least likelihood of trouble.

    • Always: The PBX directs media streams between participants when it is able to. Set this at your own risk.

    • Non-NAT Connections Only: The PBX redirects media streams only between participants who are not behind NAT.

    • Always, Using the UPDATE method: This is the same as the Always setting, but uses a different mechanism within the SIP protocol. Only set this if you have been explicitly instructed to do so.

    • Non-NAT Connections Only, Using the UPDATE method: The PBX redirects media streams only between participants who are not behind NAT, using a different mechanism within the SIP protocol.

• Always Send Early Media. Setting this option to Yes causes Switchvox to send RTP audio before a call is answered. Enabling this option may resolve an issue where audio is not sent by certain SIP providers when an inbound call is cascaded out of the same provider.

• Voicepulse Connect DID Workaround. This option is a workaround for SIP providers that send two INVITE messages for every phone call to handle SIP devices that handle DIDs and ones that do not. Only enable this option if you are sure your provider is sending two INVITEs for every call: one with the DID in the From header, and one with your account ID in the From header.

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The Fax settings determine how faxes are handled with this provider. You must install a fax license and the fax software to make faxing available.  (For details, see the “Digium Addon Products” article.)

• Error Correction Mode for G711 Faxes. The default is to leave this feature enabled.

• Minimum transfer rate for fax transmissions. Minimum transfer rate used during fax rate negotiation. The default minimum transfer rate is 2400.

• Maximum transfer rate for fax transmissions. Maximum transfer rate used during fax rate negotiation. The default maximum transfer rate is 14400.

• Maximum expected T38 packet delay. The result of this specification is that all T38 packets are padded to prepare the fax stack for the specified delay. As long as the packet delay is less than or equal to the specified value, the fax stack will not realize that there is an actual packet delay because of the packet padding. The default value of 800 milliseconds should cover most use cases.

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• IAX Provider Name. Any string to identify this VoIP provider. This is only used for referring to this VoIP Provider in other areas of the Admin Suite.

• IAX Provider Label. Label this provider gives itself when placing calls to your system. This is usually the name inside of [ ] brackets in Asterisk configuration examples.

• Your Account ID. Username provided by IAX Provider to authenticate your Switchvox PBX to the provider during IAX registration and outbound calls.

• Your Password. Your password for the account.

• Hostname/IP Address. Primary hostname or IP address of your provider's IAX

• serverServer. This hostname will be used to send registration requests. It will also be the host that will send any inbound calls to your PBX. If your provider has multiple servers for their IAX service, you can use the Primary, Secondary, and Tertiary Outbound Host settings to list them. Otherwise, the Hostname/ IP Address field will be used for placing outbound calls as well as inbound calls and registration requests.

• Callback Extension. The extension to ring when a voice call comes in over this provider or channel. Callback extensions can be any extension type (your receptionist’s phone, an IVR, etc.).

• Incoming Authentication Type. Select one of these choices from the dropdown: MD5, plaintext, and RSA. MD5 and plain text use your password for authentication. It is preferable to use the MD5 method over the plain text method. RSA uses public and private encryption keys to authenticate the provider. If you select this option, enter the public RSA key in the RSA Authentication Key field.

• RSA Authentication Key. Select from the dropdown of available keys, or from your computer hard drive.

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• Caller ID Name / Caller ID Number. Enter a name and phone number to display as your Caller ID when making outbound calls with this provider.

Connection Settings Tab

The Connection settings determine how the connection is established with this provider.

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The Fax settings determine how faxes are handled with this provider. You must install a fax license and the fax software to make faxing available. (For details, see the “Digium Addon Products” article.)

• Error Correction Mode for G711 Faxes. The default is to leave this feature enabled.

• Minimum transfer rate for fax transmissions. Minimum transfer rate used during fax rate negotiation. The default minimum transfer rate is 2400.

• Maximum transfer rate for fax transmissions. Maximum transfer rate used during fax rate negotiation. The default maximum transfer rate is 14400.

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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 change it at your provider’s request.rel6.0