Recording only for 1st port

CLI>sky rec on [filename]

Recorded file at:
/tmp/filename.pcm

Debug mode DTMF open

Debug for VISIP
/etc/asterisk/logger.conf
messages => notice,warning,error,dtmf

logs saved at:
/var/log/asterisk/messages


Debug for E1
/etc/asterisk/logger.conf
messages => notice,warning,error,dtmf

logs saved at:
/var/log/asterisk/messages

G.729 & G.723 codec translation

1. G.723 and G.729 Click Here to download

2. Copy to:
cp codec_723.so codec_729.so /usr/lib/asterisk/modules/

3. change properties:
chmod 755 codec_g723.so
chmod 755 codec_g729.so
chown root codec_g723.so
chown root codec_g729.so
chgrp root codec_g723.so
chgrp root codec_g729.so

4. restart the asterisk or whole system

5. recheck:
sudo asterisk -rx core show translation

Connection Guideline of VIT1/E1 and PBX E1/T1 module

Connection Guideline of VIT1/E1 and PBX E1/T1 module

I.    Pinouts of VIT1/E1:

Pin	Signal
1	Rx, ring, -
2	Rx, tip, +
4	Tx, ring, -
5	Tx, tip, +

II.    Straight cable pinout:

1.     RJ48/45 (VIT1/E1) to RJ48/45 (PBX E1/T1 module)

Pin (on VIT1/E1)	Pin (on PBX E1/T1 module)	Signal
1	1	Rx, ring, -
2	2	Rx, tip, +
4	4	Tx, ring, -
5	5	Tx, tip, +
3	3	Shield/return/ground
6	6	Shield/return/ground

2.     RJ48/45 (VIT1/E1) to DB-15 (PBX E1/T1 module)

Pin (on VIT1/E1)	Pin (on PBX E1/T1 module)	Signal
1	11	Rx, ring, -
2	3	Rx, tip, +
4	9	Tx, ring, -
5	1	Tx, tip, +
3	4	Shield/return/ground
6	2	Shield/return/ground

 

III.   Crossover cable pinout:

1.     RJ48/45 (VIT1/E1) to RJ48/45 (PBX E1/T1 module)

Pin (on VIT1/E1)	Pin (on PBX E1/T1 module)	Signal
1	4	Rx, ring ,- === Tx, ring, -
2	5	Rx, tip, + === Tx, tip, +
4	1	Tx, ring, - === Rx, ring, -
5	2	Tx, tip, + === Rx, tip, +
3	3	Shield/return/ground
6	6	Shield/return/ground

2.     RJ48/45 (VIT1/E1) to DB-15 (PBX E1/T1 module)

Pin (on VIT1/E1)	Pin (on PBX E1/T1 module)	Signal
1	9	Rx, ring , - === Tx, ring, -
2	1	Rx, tip, + === Tx, tip, +
4	11	Tx, ring, - === Rx, ring, -
5	3	Tx, tip, + === Rx, tip, +
3	4	Shield/return/ground
6	2	Shield/return/ground

T1/E1 PinOuts

Connection Guideline of VIT1/E1 and PBX E1/T1 module

Pin	Signal
1	Rx, ring, -
2	Rx, tip, +
4	Tx, ring, -
5	Tx, tip, +

 

II.    
Straight cable pinout:

1.     
RJ48/45 (VIT1/E1) to RJ48/45 (PBX E1/T1 module)

Pin (on VIT1/E1)	Pin (on PBX E1/T1 module)	Signal
1	1	Rx, ring, -
2	2	Rx, tip, +
4	4	Tx, ring, -
5	5	Tx, tip, +
3	3	Shield/return/ground
6	6	Shield/return/ground

2.     
RJ48/45 (VIT1/E1) to DB-15 (PBX E1/T1 module)

Pin (on VIT1/E1)	Pin (on PBX E1/T1 module)	Signal
1	11	Rx, ring, -
2	3	Rx, tip, +
4	9	Tx, ring, -
5	1	Tx, tip, +
3	4	Shield/return/ground
6	2	Shield/return/ground

 

III.   Crossover cable pinout:

1.     
RJ48/45 (VIT1/E1) to RJ48/45 (PBX E1/T1 module)

Pin (on VIT1/E1)	Pin (on PBX E1/T1 module)	Signal
1	4	Rx, ring ,- 1 Tx, ring, -
2	5	Rx, tip, + 1 Tx, tip, +
4	1	Tx, ring, - 1 Rx, ring, -
5	2	Tx, tip, + 1 Rx, tip, +
3	3	Shield/return/ground
6	6	Shield/return/ground

2.     
RJ48/45 (VIT1/E1) to DB-15 (PBX E1/T1 module)

Pin (on VIT1/E1)	Pin (on PBX E1/T1 module)	Signal
1	9	Rx, ring , - 1 Tx, ring, -
2	1	Rx, tip, + 1 Tx, tip, +
4	11	Tx, ring, - 1 Rx, ring, -
5	3	Tx, tip, + 1 Rx, tip, +
3	4	Shield/return/ground
6	2	Shield/return/ground

How to play and record voice by using Lexis30

安裝好lexis30並且將30路啟動

更改讀取權限(否則這兩個資料夾無法讀取)
sudo chmod -R 777 /var/lib/asterisk/sounds
sudo chmod -R 777 /var/spool/asterisk/monitor

將要撥放的檔案放入 /var/lib/asterisk/sounds
錄好的檔案會放在 /var/spool/asterisk/monitor

在extensions.conf裡新增一個context[testnewout]
skype端錄音配置成
exten => _X.,1,Wait(9)
exten => _X.,n,SendDtmf(${EXTEN})
exten => _X.,n,NoOp(${EXTEN})
exten => _X.,n,Wait(1)
exten => _X.,n,Monitor(wav,${EXTEN})
exten => _X.,n(play),Wait(10)
exten => _X.,n,Goto(play)

skype 放音 配置成
exten => _X.,1,Wait(9)
exten => _X.,n,SendDtmf(${EXTEN})
exten => _X.,n,NoOp(${EXTEN})
exten => _X.,n,Wait(1)
exten => _X.,n(play),Playback(語音文件名)
exten => _X.,n,Goto(play)

將test2.c放置於任一資料夾(例如test) 按此下載
compile 成一個檔案(例如testapp)
sudo gcc test2.c -o testapp

撥打的命令
在 testapp所在的目錄下 sudo ./testapp X skypeID (分機號)
X代表共幾路呼叫,SkypeID可填入代表號,分機號則填入欲撥打的分機號
例如 sudo ./testapp 1 voskytp1 325為1路撥打voskytp1後轉分機325

使用這個來錄音，先在extensions.conf裡依照上面錄音端設定
然後依以下的指令來撥打
sudo ./testapp 1 tpae01 991
sudo ./testapp 2 tpae02 992
sudo ./testapp 3 tpae03 993
依此類推，這樣子會在放錄音檔的地方產生991-in.wav,9910-out.wav,992-in.wav....

使用這個來放音，先在extensions.conf裡依照上面放音端設定
然後依照以下的指令來撥打
sudo ./testapp 1 tpae01 991
sudo ./testapp 2 tpae02 992
sudo ./testapp 3 tpae03 993
依此類推，這樣對方相對應的分機就會接起來後就會聽到聲音
PS.放音建議可以使用windows+skype+音源線會比較快

顯示一些特別保留碼的語法

＜範本＞這是一個要顯示保留碼的測試文章＜/範本＞

使用方法為
"&#65308" + ";"(連在一起)，就可以顯示出 "<"
"&#65310" + ";"(連在一起)，就可以顯示出 ">"