SIP Client
From Yate Documentation
(Difference between revisions)
(→URI calling with Yateclient) |
|||
| (3 intermediate revisions by 2 users not shown) | |||
| Line 22: | Line 22: | ||
==Starting YateClient== | ==Starting YateClient== | ||
| − | + | 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 | ||
| Line 39: | Line 39: | ||
[[File:register_sip_account.png]] | [[File:register_sip_account.png]] | ||
| − | Select the | + | Select the SIP protocol, insert the SIP username, the SIP password and the Server IP. |
| − | + | ||
==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. | + | 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. | + | |
| + | The SIP destination has the following format: sip/sip:_username_@_ip_address_:port. | ||
| Line 53: | Line 53: | ||
* [http://yateclient.yate.ro/index.php/UserGuide/CallingwithSIP Calling using SIP] | * [http://yateclient.yate.ro/index.php/UserGuide/CallingwithSIP Calling using SIP] | ||
* [[Telephony]] | * [[Telephony]] | ||
| + | |||
| + | [[Category:SIP]] [[Category:Client]] [[Category:YateClient]] | ||
Latest revision as of 10:57, 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
