1. Fail2ban - 這個不是Asterisk 內置，但好有用。
2. Google Voice - 用電話機直接打出、接入 Google Voice 電話，真係方便到笑咗出黎，這功能我等咗大半年，以前要用Web 撥打，很少用，現在太好了。
3. Google Calendar - 未 set，稍後再試。作者: Qnewbie 時間: 2011-1-4 19:06
For google voice, the direct dial through gizmo5+sipsorcery(cannot register for free now) has been used for a while.
Next week, when the IP01 is arrived. The migration to IP01 beginz. The atom-PC might used for other objective, like skype trunk using siptosis.作者: bubblestar 時間: 2011-1-4 19:41
I could not use direct dial as I have been using Sipgate instead of Gizmo5, hence, not aware of this changes. BTW and for you info, Asterisk 1.8 no longer supports dialing from its Google Web. Despite of that, Asterisk 1.4 can still use this traditional method.
Speaking of Skype trunk, this can be a good topic for me to learn. If it rings a bell to those who are familiar to this, please give us a hand.作者: ckleea 時間: 2011-1-4 20:01
Just US$23 you can get the program and trunk builder. Mine is very old. Just US$9.99 for both作者: Qnewbie 時間: 2011-1-4 21:25
For the free siptosis, installed in Ubuntu(not atom PC), it responses a bit slow. Perhaps the pay-version would be better.
A lot of callback service with Skype as backbone can be found in taobao. Might I make a query to ckleea C-hing, could pay-version siptosis be used for this end?作者: ckleea 時間: 2011-1-4 21:36
I am not sure what you mean. For my setup, the quality is just the same as usual VOIP or PSTN line.
If you look at this for the changes, it just worths the money spent
Mine is still 2009 Oct one作者: ckleea 時間: 2011-1-4 22:10
剛用google voice互打，網絡起碼經兩次太平洋，同固網基本上一樣。
作者: Qnewbie 時間: 2011-1-4 22:12
Oh, the siptosis closes the skype connection 5 to 10 seconds after I hang up. That is why I said slow response.
The skype callback used by taobao seller after google searching is called QQ callback(?). The detailed info is unknown. But if a single account is involved, it should be some kind of conference to connect two PSTN users(+skype acc. self). The question might be formulated by following: could we use a single skype account to call two PSTN user and connect them together with conference with siptosis?作者: ckleea 時間: 2011-1-4 22:21
You mean you can still call gizmo5 directly from Hard Phone thru Asterisk 1.8 directly? I can't. Just hear 2 ringing and then call drops.作者: ckleea 時間: 2011-1-6 06:26
Yes, I still use Gizmo5 to call but I do not register their server all the time. Only when needed.作者: bubblestar 時間: 2011-1-6 11:13
建議如下，
register string at sip.conf to remark out not to use，即加";"
the remaining gizmo5 dial out plan reserved at sip.conf so that when needed, can dial out作者: bubblestar 時間: 2011-1-6 12:13
Thanks for your recommendation. Such arrangement can avoid editing much work from both sip.conf and extensions.conf. Only change a single sip.conf file can simplify our administration.作者: Qnewbie 時間: 2011-1-6 22:55
WOW, never think that the gateway function is available for asterisk too. SPA3000 has this function but I did not test it.
Could ckleea or bubblestar explain a little more on this kind of on-the-fly registration?作者: ckleea 時間: 2011-1-7 05:56
