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 ">"

Skype Codecs

1. C:\Documents and Settings\account\Application Data\Skype\account\config.xml



2. [forcecodec]G729[/forcecodec]



3. [disablecodecs]SVOPC SVOPC_SB AMRWB[/disablecodecs]



Example:

[call]

[forcecodec]G729[/forcecodec]]/forcecodec]

[/call]

Restart Zaptel setting without restart system

After changing setting of Zaptel

Only restart Asterisk is not enough

you can type "sudo ztcfg -vv" to restart the zaptel driver to make you setting take effect

Asterisk adjust gain

How to tune the volume in asterisk system



configure zapata.conf



rxgain represents the receive volume

txgain represents the transmit volume


Default value is 0.0


PS:Most reading suggests going no lower than -11.0 and no higher than 11.0, though it will take values from -100 to 100.Make sure that rxgain=/txgain= lines are placed prior to channel= line in your zapata.conf otherwise the gain settings will not have any effect.



Reference link:Click Here