SIP Client

Latest revision as of 11:57, 4 November 2013

Yate can be used as a SIP Client. An implementation for this is Yate Client.

As a SIP Client, Yate has the following features:

• Voice - implemented
• Video - soon to be implemented
• All free codecs - gsm, speex, alaw, mulaw
• Registration - implemented
• Autentification to multiple endpoints - implemented
• Multiple lines (channels) - implemented
• Voicemail - depending on the service
• Hold - implemented
• Transfer - implemented
• Conference - implemented
• ENUM - soon to be implemented
• Dialing using URI - implemented
• Smart routing - low cost routing via multiple providers - implemented
• TFTP provisioning - soon to be implemented
• Local address book

[edit] Starting YateClient

If you are using a Linux platform, after installing go to 'clients' directory from yate sources and run:

./run-qt4 -vvvvv -CDo

• -v: enable levels for debugging
• -C: enable core dumps if possible
• -D: special debugging options
• o: colorize output using ANSI codes

[edit] Register SIP account in YateClient

If you have a registered SIP address/account with a SIP provider, you can register this account in YateClient.

From Yate Tab select 'Add account' and a window will open like in the next image:

Select the SIP protocol, insert the SIP username, the SIP password and the Server IP.

[edit] URI calling with Yateclient

In YateClient, in Telephony Tab, manually type the exact SIP address at which the other party can be reached.

The SIP destination has the following format: sip/sip:_username_@_ip_address_:port.

See also

• YateClient
• Calling using SIP
• Telephony