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標題: Yate Client [打印本頁]
作者: ckleea    時間: 2011-2-7 21:18     標題: Yate Client

本帖最後由 ckleea 於 2011-2-7 21:32 編輯

I found this client is quite good for us. It allows multiple SIP and IAX registrations.

http://www.yate.null.ro/pmwiki/index.php?n=Main.YateClient

It can be used as an alternate to Zoiper especially that if you don't want to pay.

圖片附件: screenshot.07-02-2011 21.23.35.jpg (2011-2-7 21:32, 24.66 KB) / 下載次數 503
http://telecom-cafe.com/forum/attachment.php?aid=446&k=1c7e68b17204d958fad3ee5be26eda0c&t=1732292320&sid=XX6k8L

作者: Qnewbie    時間: 2011-2-7 22:57

It is claimed that GV is included...

http://yate.null.ro/pmwiki/

31 Jan 2011:
Yate 3.1 released.
Yate client support for Google Voice. Support for any country tones in tonegen.
作者: ckleea    時間: 2011-2-8 20:28

回復 2# Qnewbie


    I don't think so for support of google voice in 3.1.
作者: Qnewbie    時間: 2011-2-8 21:08

I don't know since I dial GV thru sipsorcery. But the guide does mention the GV.

http://yateclient.yate.ro/index.php/UserGuide/CallGVoice

Cheers!
作者: ckleea    時間: 2011-2-8 21:32

回復 4# Qnewbie


    Yes, you are right. I overlook this.

BTW, how to configure SIP Sorcery?
作者: Qnewbie    時間: 2011-2-8 22:34

I just followed the wiki document:

http://code.google.com/p/google-voice-sipsorcery-dialplans/w/list
作者: ckleea    時間: 2011-2-9 07:03

回復 6# Qnewbie

Thanks. Looking interesting. I am trying to make it for me.

Are you putting sipsorcery into a sip trunk for IP01?

Could you put out your example.
作者: Qnewbie    時間: 2011-2-9 19:10

I use ipkall as dial-in number, which does not need gizmo5 number. The detailed setup at the sipsorcery can be found in the wiki.

The IP01 end is just as normal sip trunk.
  1. Provider Name  : sipsorcery       
  2. Hostname : sip.sipsorcery.com
  3. Username  : your_sipsorcery_extension_name       
  4. Authuser        : your_sipsorcery_extension_name       
  5. Fromuser  : your_sipsorcery_extension_name                
  6. Fromdomain  : sipsorcery.com       
  7. Password : your_password
複製代碼
Other setups like inbound and outbound rules, dialplan etc., would be modified as normal sip trunk.



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