SIP Client
From Yate Documentation
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* [http://yateclient.yate.ro YateClient] | * [http://yateclient.yate.ro YateClient] | ||
| + | * [http://yateclient.yate.ro/index.php/UserGuide/CallingwithSIP Calling using SIP]] | ||
Revision as of 16:10, 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
