UserGuide: Calling using SIP


A. Calling using SIP - while logged into a telephony account

A1. Log into your SIP account
A2. Make the call to a SIP destination

B. Calling using SIP - URI calling with YateClient - not logged in

B1. Adding SIP address from Telephony Tab
B2. Type the destination you want to reach

If you want to make calls with YateClient to a SIP account or a destination, you can do that in two very distinct ways - let's say:

While logged into a telephony account

A. You have registered for a SIP address/account with a provider that offers SIP telephony services. In this case, that specific provider will take care and handle your call.

To place a call using your existing SIP account, do the following:

A1. Log into your SIP account:
1. Use the wizzard presented here or click directly Yate -> Add account.

Choose the appropriate SIP services provider from the list. If you cannot find it, leave it with "none"

Protocol: SIP

Username: your_sip_account_username

Password: your_sip_account_password

Server: Complete here if you didn't find the provider in the first list, Use provider list.

2. To check the status of your account, you can go to: Settings -> Accounts. Here is a list of all your account, with the appropriate status (the checkbox is ticked when the account is enabled).

A2. Make the call to a SIP destination.
Go to the telephony tab, and choose the "Account" from which to make the SIP call.
Advanced mode - unchecked


You have have only "Account" to choose from.
If you have just one account that you are logged in, this part is missing.

Advanced mode - checked


You have the possibility of choosing "Protocol".

A3. Using the dialpad or directly the numerical keys, input the number you want to call.


Calls button. Right now you should begin your call:

That's it!

URI calling with YateClient - not logged in

B. You manually type the exact SIP address at which the other party can be reached, like in the following steps:

B1. Go to telephony tab and press the

Calls button:

B2. Type the destination you want to reach in the following format: sip/sip:_username_ @ _ip_address_:port
For example:
Protocol Slash part of URI syntax Username

can be a number

at IP colon Port Whole example
sip / sip: 506 @ : 5060    sip/sip:506@

If you want to read more about how is SIP protocol implemented in Yate as a telephony engine, see here.

  Screenshots are from YateClient version 4.0.

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Page last modified on January 28, 2013, at 01:42 PM