SIP Client
From Yate Documentation
(Difference between revisions)
| Line 22: | Line 22: | ||
* [http://yateclient.yate.ro YateClient] | * [http://yateclient.yate.ro YateClient] | ||
| − | * [http://yateclient.yate.ro/index.php/UserGuide/CallingwithSIP Calling using SIP | + | * [http://yateclient.yate.ro/index.php/UserGuide/CallingwithSIP Calling using SIP] |
Revision as of 17:10, 8 November 2012
Yate can be used as YateClient and has the following functions:
- 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
See also
