SIP Client

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==Starting YateClient==
 
==Starting YateClient==
  
After [[Installation|installing]] Yate, if it way installed on linux platforms than you can go to where the yate sources are in directory 'clients' and run:
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If you are using a Linux platform, after [[Installation|installing]] go to 'clients' directory from yate sources and run:
  
 
  ./run-qt4 -vvvvv -CDo
 
  ./run-qt4 -vvvvv -CDo
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[[File:register_sip_account.png]]
 
[[File:register_sip_account.png]]
  
Select the sip protocol, insert the SIP username, the SIP password and the Server IP.
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Select the SIP protocol, insert the SIP username, the SIP password and the Server IP.
 
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==URI calling with Yateclient==
 
==URI calling with Yateclient==
  
In YateClient, in Telephony Tab, manually type the exact SIP address at which the other party can be reached.  
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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.
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The SIP destination has the following format: sip/sip:_username_@_ip_address_:port.
 
   
 
   
 
   
 
   
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* [http://yateclient.yate.ro/index.php/UserGuide/CallingwithSIP Calling using SIP]
 
* [http://yateclient.yate.ro/index.php/UserGuide/CallingwithSIP Calling using SIP]
 
* [[Telephony]]
 
* [[Telephony]]
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[[Category:SIP]] [[Category:Client]] [[Category:YateClient]]

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:

Register sip account.png

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

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