Modify Providers

Provider Types:

There are 2 types of providers when you are a VSP, these are "Termination Providers" and "Online VSP's":

1) "Termination Providers" - Termination providers are providers who have their own equipment physically connected to the PSTN (POTS) network through a provider such as Verizon, Bell, Telstra, Optus, Vodafone, Etc. These providers generally provide Online VSP's with the ability to wholesale their termination points (where the phone network connects to their equipemnt).

2) "Online VSPs" - Online VSPs are VSP's which have a presnce online only. Such as a server or server cluster hosted in a data center which they intend the general public to access. Online VSPs do not have equipment connected to the PSTN (POTS) network and thus rely on termination providers to take their customers call that final leg into the public telephone network.

What am I?

vspPanel allows you to be either. If you are intent on setting up your own terminations you will however need more knowledge of core asterisk and the Digium hardware available to make the connections. Once you have made your termination points vspPanel will allow you to connect to them and start routing your customers as if you were an online vsp.

The most common use of vspPanel is to manage online vsp's however as these can be setup quickly and usually present a saving on terminating your own traffic unless you have at least several million minutes of voice traffic to pass through a month.

What is SIP and IAX?

SIP can be regarded as the enabler protocol for telephony and voice over IP (VoIP) services. The following features of SIP play a major role in the enablement of IP telephony and VoIP:

Name Translation and User Location: Ensuring that the call reaches the called party wherever they are located. Carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported.
Feature Negotiation: This allows the group involved in a call (this may be a multi-party call) to agree on the features supported recognizing that not all the parties can support the same level of features. For example video may or may not be supported; as any form of MIME type is supported by SIP, there is plenty of scope for negotiation. Call Participant Management - During a call a participant can bring other users onto the call or cancel connections to other users. In addition, users could be transferred or placed on hold.
Call feature changes: A user should be able to change the call characteristics during the course of the call. For example, a call may have been set up as voice-only, but in the course of the call, the users may need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call.
Media negotiation: The inherent SIP mechanisms that enable negotiation of the media used in a call, enable selection of the appropriate codec for establishing a call between the various devices. This way, less advanced devices can participate in the call, provided the appropriate codec is selected.

Where as the IAX revision 2 protocol is used by the Asterisk VOIP PBX and FreeSwitch Softswitch as an alternative to SIP, H.323, etc. when connecting to other devices that support IAX (a limited list at the moment, but growing very rapidly). However meanwhile other software like Yate SofaSwitch and OPAL have added IAX2 support.

IAX is not the result of a standards group, rather a collaborative, community based effort.
IAX uses a single UDP port 4569, and thus works well in NAT environments (the obsolete IAX1 protocol used port 5036). IAX uses ONLY one udp port for both control and data traffic. With IAX you will always have audio if the control connection can be established.
IAX supports PKI-style authentication and trunking.
The Asterisk IAX2 driver has a jitter buffer. (The SIP driver doesn't - yet).
Once you have selected the protocol that you want to use (make sure the provider supports this protocol), simply click on the “Next” button.

Adding a Provider

When adding a provider you will need to choose between SIP and IAX as the providers protocol. Your provider should give this information too you and you should simply choose the correct one.

After choosing the providers protocol, you will then be asked to enter in some settings relating to that provider. The provider should also supply you with these.

The following is an explanation of the settings.

Provider Name – This is the name of the trunk connecting to the provider, you may only use alphanumeric characters, NO SPACES.

Host – This is either the domain name or IP address of the provider’s server you are going to connect to.
Username – Is the username that the provider has given you to connect to their servers.
Password – Is the password that has come with the username from the provider.
Authentication Method – Is the encapsulation method used to connect to the provider, at the moment the only selection is MD5, other options will come up when they are added.
Type – Is the method in which you will connect to your provider, for all providers at the moment this should be set to “peer”.
The advanced options section is only optional:
Caller ID – This is the outgoing caller ID option, if your provider allows for it all outgoing calls will be set to this CID.
Qualify – Checks if the client is reachable, if yes the checks occur every 2 seconds
No Transfer – This option checks if the connection should re-connect if the connection is lost to the providers end-point, this should normally be set as yes so it will not transfer.

The following codecs section is a list of codecs and bandwidth usage. Depending on what bandwidth limitations you have you should choose the most appropriate connection codec. Please note that some providers do limit codecs so check with your provider for allowed options.

GSM - 13 Kbps (full rate), 20ms frame size
iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
G.711 - 64 Kbps, sample-based Also known as alaw/ulaw
G.726 - 16/24/32/40 Kbps
G.729 - 8 Kbps, 10ms frame size

Speex - 2.15 to 44.2 Kbps

After you have entered in all the above settings just click on the "Add New Provider" button

Example 1 - From Getting Started With vspPanel

Lets setup my first provider, who can I use?

Setting up your first provider is really easy in vspPanel. you can use ANY SIP or IAX based provider which means you really have your pick of them. Since vspPanel was designed by FaktorTel lets use them as an example. Here's the steps:

1) Go to: www.faktortel.com.au

2) Signup for a FREE account from the main page, using the option which says you have your own phone and only want a username / password.

3) Once you receive your username and password open your vspPanel and click on "Modify Providers"

4) Click on the big world with the plus symbol on it to add a new provider.

5) As FaktorTel Uses IAX choose the IAX protocol. (For SIP providers you would choose SIP) and click on Next

6) Now, you should have been provided with a Username and Password and SIP / IAX address for the provider. Enter these in the appropriate fields. For FaktorTel these would be:

Provider Name: FaktorTel

Host: iax.faktortel.com.au

Username: Your Username

Password: Your Password

Now Leave the rest as per default....

6) Press "Add New Provider"

7) Your provider should be added and working. At the moment you can only check if you are registered through the Asterisk Console. However as long as you have entered in the correct information you should be connected to the provider immediately. (No reloads necessary).