VoIP Account Name

This is the name of the VoIP account. This field will be passed on as part of the caller ID information when a call is made using this account.

Number of simultaneous lines allowed

This defines how many instances of the VoIP Account you would like to be created.

Note that if the provider of your account only allows one instance to be configured then setting this number higher than "1" will be of no benefit. Check with your SIP provider for the number of simultaneous calls you can use with your account. If the account is an Axon extension it can be used as many times as desired.

SIP Number (or User Name)

This is the user name provided by the VoIP gateway service. In most cases it is the actual phone number assigned to you for this account.

Password

The password for your VoIP account.

Server (Proxy and Domain)

This is the IP address or domain name of the server through which your account is provided. If you were not given this information in an email or webpage, you must ask your provider what it is.

Outbound Server

Some users may need to use an outbound proxy server to get SIP requests to work correctly through a firewall or with legacy equipment. These can include a simple SBC (Session Border Controller), a SIP enabled firewall router or even a media transcoder to allow connections to legacy VoIP equipment. If you are running a SIP application that connects to a SIP PBX on the local network (LAN), then you will not need to use an outbound server. If, however, you need to pass a request through a computer on the DMZ of your network (if you don't know what this is you probably don't need to do this), this is generally done to pass requests through the firewall through a single centralized gatekeeper program called an outbound proxy server - you may need to enable the outbound server option. Some SIP providers may also require a different outbound server on their network. This option can be used to enable that as well.

These settings relate to your computer's ability to make phone calls over a network or the Internet. Most of the options here are fairly advanced, so don't change them unless you understand them fully.

Local SIP port

The UDP port on your computer that will be used to create a SIP signaling connection. If the port is in use then the software will find the next available port.

Local RTP port

The UDP port on your computer that will be used to for the audio connection. If the port is in use then the software will find the next available port.

NOTE You must open both of the above local ports on your local computer's firewall. Your router should be able to handle opening its own ports automatically, but if it can't then you may need to setup port forwarding.

Use UPnP to find external IP address

Uses the Universal Plug 'n' Play (UPnP) protocol to determine your external IP address. It is a quick and easy way of having your router configure itself. Many routers have UPnP disabled by default so you might have to enable UPnP on the router first. Refer to your router documentation for instructions.

Some routers are unable to use UPnP at all and some older routers do not do it effectively. If you have one of these routers, enabling UPnP may cause problems with outbound dialing and with voicemail.

Note that if you are using Microsoft Internet Connection Sharing (ICS) over a LAN, you may need to enable UPnP in Windows. To do this, on the computer running Internet Connection Sharing select Start -> Settings -> Control Panel -> Network Connections and then right click on the Internet connection and select Properties. Go to the Advanced tab and check the option "Allow other network users to control or disable the shared Internet connection".

Use STUN servers to find external IP address and port

Uses the "Simple Traversal of UDP protocol through NAT" technique to ask a server on the Internet if you are behind a NAT or firewall, and if so, what your external IP address is. The default STUN servers should only be changed if your SIP provider has given you a different STUN server to use.

Use static IP address and static mapped ports (Advanced Option): This is where you can enter your external IP address and SIP/RTP ports if your computer, or the server computer on which your network connects to the Internet, is using a static IP address. If you choose to use this option then we recommend the SIP and RTP ports are set to 5060 and 8000 respectively.

If you are having problems making phone conversations and suspect the issue is network-related, then click the "Run Network Setup Wizard" button in order to launch a wizard that will guide you through the network troubleshooting process.

These settings relate to the choice of codec when making a phone call. There is an inverse relationship between the quality of audio and the amount of bandwidth that will be required. If you want to keep your bandwidth down which can be useful for slow internet connection, the codec used will result in a lower quality audio. Alternatively if you are looking for higher quality audio you can select the codec for high quality audio which will require using more bandwidth.

Answer OGM

By default, all lines answer starting with the default Out-Going Message (OGM) that is set in the General System Settings in the "System Settings" tab on the web interface. If you are running multiple lines, and want to override this default so different lines have different starting OGMs, use this drop down list to select the OGM that will override the default for this line.

Answer Mode

You can also override the default 'Answer Mode' that are set in the General System Settings by selecting the 'Override Default Options' check box. These options allow you to set the number of rings or a tollsaver option for individual lines, changing the number of rings or the tollsaver options as appropriate. If at any time you wish to go back to using the default values simply uncheck the 'Override Default Options' box and of course click "save changes" at the bottom of the window.

Disable Caller ID Flash for this line

If you are running multiple lines, but want Caller ID Flash/Announcement on only some lines, you can use this check box to disable CID Flash on the line. For more information see Caller ID Blocking.

Allow outbound calls

By selecting 'Allow outbound and message forward calls on this line' IVM will use this line when making outbound or mailbox message forward calls. Normally all lines will have this selected. If you have multiple lines and one is a priority incoming line that you do not want to be busy you can turn off outbound calls on the line. You may also want to limit the lines that are used by the automated outbound dialer; when making outbound calls, IVM searches for a free line with outbound dialing enabled, searching through the lines in the order the lines were added.

For more information see Outbound Autodial Calls. Also see Forward to Telephone Number Message notification in General Mailbox Properties

SIP Message Logging

Log all SIP messages This will log all the SIP messaging data into a file.

Log all RTP (audio) packets This will log all audio data that is sent and received.

SIP and RTP data are both stored in the one file for convenience. Files are split up according to day in order to avoid creating large files. This option is used when you are experiencing difficulties with the software and need to either look at a SIP message log or need to send a log to NCH to assist problem resolution. This option should be left off unless needed, as it will write data to files which can potentially slow down the software. Once enabled, this option can only log future calls, it cannot log what has happened beforehand, so it must be enabled before running any call testing scenarios.

Logs can be viewed from the "Logs" tab. See IVM Log Files for more information.