If you need to add a new internet telephony service account to your provider list, please choose "Settings" in Start > All Programs > AGEphone Business. Once the Settings screen pops up, press the “Add new” button. Enter your public (SIP) address into the displayed dialog box. This information is supplied your IP telephony service provider when you sign up for an account. The format is as follows: “User Name @ Provider Domain.tld.”
Choose the service provider you would like to use3 from the provider list. The selection should now be highlighted. Click the “Set as Default” button in order to designate your outgoing calls to use this service provider. Set a checkmark next to the server name to make AGEphone Business register to this server so that you can receive incoming calls.
The list of providers is displayed under the input area. All addresses have a box to the left respectively and you can select which service to use by simply clicking that box. In other words, the providers that have not been selected will not be able to receive calls.
Moreover, the IP phone service set as default by the user is the one used to place calls until the default is changed or another provider selected via the Provider Quickswitch feature.
“SIP server-less” comes in handy if you would like to make calls without using any server, instead opting for a peer to peer conversation between two devices over LAN or WAN. If enabled, all you need to do is input your partner’s IP address.
Some of the information on this screen will be automatically derived from the Public address entered in the previous step. However, please make sure to fill in the blanks as required by your provider. In this sense it is worth noting that that Authorization ID and User ID can be different sometimes.
In this window, advanced settings for the server can be entered. Input a phone number in the “Display Name” area if you would like to place calls using that number as your caller ID (only if supported by your provider). If you do not have a dial in number or if you do not wish to use it, you can enter any character string or simply repeat the “User Name”.
The “Authorization ID” is intended for servers that require a separate authentication and can (but doesn’t have to) be different from the “User Name” contained in the SIP address. The password is always the same for both IDs.
The Realm is usually the same as the Provider Domain used in the public address string. There are numerous exceptions and it's best to consult your provider if you are unsure and the information hasn't been published on their page. The Realm also determines under which name your VoIP service shows up in the provider list.
The input area for the SIP Server address determines the actual machine that the connection is made to. Some service providers call this field “SIP Proxy” and also use this address as the Outbound Proxy when calls are placed. As the SIP Server and SIP Registrar address are different if there is a separate Outbound Proxy, this is a simplification that makes it impossible to get some services to work. AGEphone Business on the other hand is prepared for all eventual configurations.
Required Media Type is used to enforce a certain voice codec for incoming calls. The numbers stand for (3) GSM-FR, (0) G.711.u and (8) G.711.a. Setting this value can be important if you have to conserve bandwidth or if your internet telephony provider has problems with one of the above codecs.
(Please be aware that transfer may fail if the Media Type (payload type) list for receiving incoming call is not supported Media Type at forwarding destinations’ server). 0:G.711ulaw 3: GSM 8:G.711alaw
The Dial Prefix is required for cases when you would like to prepend a string (not just numbers) in front of your dialed number. The counterpart to the Dial Prefix is the Dial Suffix which appends any string you enter after the dialed number. Both fields are not required for making the provider work with AGEphone Business!
The country code field allows you to insert any country code for each number dialed through this provider. This is practical when you have stored your numbers in a local format but your VoIP provider needs them to be dialed in international format (as most do). Entering your country code in this field makes sure that it is used for each and every number. Please keep in mind that when using this feature and placing an international call that requires another country code, the call will fail because the wrong code is being used to prepend the number.
Here you can choose between two methods of DTMF tone transmission: "RFC2833" and "Inbound". Both do exactly the same, but while some VoIP providers may offer support for both variants, others respond to only to one of the methods. You will notice that you have to change the setting when calling an automated voice system and your key presses are not registered.
Symmetric Response Routing is a feature that allows NAT traversal without "STUN / UPnP" in certain network situations and if the SIP server supports it. The option is disabled by default because most SIP servers don't don’t offer this feature and even if they do there are many situations where it doesn't work reliably. If your network requires NAT traversal you are usually better off using "STUN / UPnP".
Paging is a function that reacts with certain keywords in SIP packet headers and allows e.g. to auto-answer calls if the requirements are met. In order to use this function, a server which supports paging is required. The keyword for the above example in AGEphone Business would be “auto-answer=0” which each incoming SIP packet header will be checked for.
It is usually for the best to leave these network settings alone. They allow you to switch the port management from automatic to manual for the very few cases where it is necessary. It's safe to leave this option as it is under normal circumstances as AGEphone Business is perfectly capable of picking the correct ports its own.
※This page allows the user to select the ports. As usually there is no particular reason to set the port number manually, just choose Automatic Port Selection.
These options are essential for making AGEphone Business work with different networks. Whenever your computer or device is part of a local or wide area network without a direct connection to the internet, "STUN / UPnP" is a usually best to get through to your partner.
Manual configuration can help in more complicated and non standard network setups that you know better than AGEphone Business can detect them. Just enter your local (machine <–-> router) and global (router <--> internet) addresses to get rid of such problems as one way audio. Contact your network administrator if you are unsure how to set the addresses.
For all direct internet connections "Local IP" is the best choice as "STUN / UPnP" cam cause problems at times and is at least wasting resources because it constantly sends out packets to make sure that the ports selected are still open. Depending on your choice, some options on this page will be inaccessible and grayed out.
This option enables AGEphone Business to answer on its own when there is an incoming call. In the left column you can choose immediate reactions and in the right column those that you would like to see triggered after a certain interval has passed. This interval can be set in seconds in the Incoming field. If you would like to forward the call, please make sure to select “Automatically Answer Call” from the table.
You can specify the forwarding information in the lower two fields. Please provide the destination number in the left and select the according provider in the right half.
Please make sure to use an alternative account to forward the call if your VoIP provider does not allow for more than one call at the same time. Keep in mind that the transfer will incur fees according to your provider’s terms of service.
This setting allows you to select which audio device to use if there are multiple audio devices plugged into your computer. If that is not the case, simply leave it as System Default. In the Ringtone file dialog you can select any 8 KHz, 8 bit mono .wav file as your ringtone.
Microphone Volume lets you set your outgoing volume which will always be set when starting AGEphone Business and which is switched back to the system default upon closing the softphone.
Jitter-Buffer should be set according to the quality of your connection which may introduce more or less packet fragmentation that AGEphone Business has to make up for with this buffer to prevent stuttering.
Activating Direct Sound sometimes enhances the overall sound quality with certain audio devices. If you hear treble, distortions and stuttering it might fix these problems somewhat. Please note that in the latter case it is probably better to raise the jitter buffer instead.
Voice Activity Detection stops the audio transmission when there is no voice activity, thus saving bandwidth and keeping noise on the line to a minimum. Our algorithm smoothes out the transition between the different voice transmission states and conceals it effectively.
If you set Record Calls then each and every call will be recorded into the "My Documents \ My Phone Booth \ Account Name" folder. Where exactly in your user profile “My Documents” is located depends on your Windows OS version.
Setting Launch on Startup makes AGEphone Business run on Windows start which saves you the trouble to start it manually every time. As the softphone uses but little resources, you don’t need to worry about your system getting slowed down by this.
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