DTMF in-band and out-of-band

DTMF in-band means DTMF tones are sent as audible tones along with the voice (DTMF is sent IN the voice frequency BAND).
DTMF out-of-band means DTMF tones are sent as the numerical value in a packet (RTP type telephone-event, refer RFC2833).
High-compression codecs such as G.729 are designed to carry voice waveforms not tones, so may have problems with DTMF in-band.
You can set your VOIP device either way. Whether out-of-band works depends on your provider also supporting it.

[Call features] Calling Line Identification and Prohibit

Call ID 顯示與隱藏


使用電話按鍵來控制是否顯示Caller ID

Absentee Service

This facility
enables you to attend the calls in your absence. This feature works like a Voice
Mail Service without using answering machine. If you avail and activate this
facility, you will not receive any incoming call on your Phone, the tape will
indicate your absentee to the calling party. During this period, outgoing calls
can still be made as usual. On lifting the handset a different dial tone will be
heard indicating that the Absentee facility is activated. For this service,
calling party's meter will not operate. You can activate or deactivate this
facility using following codes according to your convenience.

關於Ubuntu

Ubuntu的Alternate CD能夠在極少的硬體資源下安裝Ubuntu到電腦裡，因為不執行Live CD，所以不會使用太多運算。

使用VoIP做任何的事

看到了一篇很有趣的文章，他的標題是如何:使用VoIP做任何的事

當然也不是任何事啦，但他舉了一些很有趣的應用。



1.Call Mapping(Gnome feature)

在Google地圖上秀出發話位置



2.Caller ID on TV(Skype addon)

在電視上秀出來電提示



3.Transferring Voicemail to Email(Asterisk feature)

將語音留言寄到Email



4.Call Recording(當通話端是電腦的時候)

錄下通話內容



5.Call Encryption(zfone對電腦端VoIP client加密)

將通話內容加密



6.Document Editing(Skype addon)

協同處理文件



7.Desktop Sharing(Skype addon)

共享桌面



8.Bluetooth Integration

使用藍芽耳機麥克風



9.Conventional VoIP Phones

10.Nintendo DS VoIP phone

11.Virtual Tutoring

12.Voice Alteration





進階使用

1.Wakeup call(Skype)

讓skype在特定時間撥電話給你



2.Listen to Podcast.(Asterisk)

使用Asterisk聽Podcast



3.Torture Telemarketers.





對於幾項關於Asterisk跟Skype應用，還蠻有興趣的。

PSP使用VoIP的服務

VoIP & Gadgets Blog,



話雖如此,會影響使用意願的還是其介面吧，想到用PSP的十字鍵來選擇電話號碼，就覺得麻煩。

相較之下，我還是比較期待在NDS觸控螢幕的加持下，還在發展的DSiP

Peer 2 Peer call 單通

phone ------PBX----- gateway(香港)-------------gateway(大陸)------PBX-------phone



今天處理了一個p2p 單通的問題

說也奇怪

單通主要是因為NAT 擋住了RTP通道的通訊而發生的

但今天兩個點都在public ip還會發生這種問題



處理方式 是把他改成sip協定

現象就消失了

看來是硬體的問題



不過好玩的是測試的方式

VoIP call-------gateway(香港)------PBX-

-----匯線通(中央交換機)

-----轉回PBX--------gatewUay(香港)-----VoIP call

等於是做了一個U-turn在測 不知道這樣會不會有問題