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標題: 香港、澳门、珠海、广州 VoIP电话系统 (QNAP TS-453 Pro + Debian 8 + Asterisk 13) [打印本頁]

作者: 角色 時間: 2015-6-5 21:38 標題: 香港、澳门、珠海、广州 VoIP电话系统 (QNAP TS-453 Pro + Debian 8 + Asterisk 13)

本帖最後由角色於 2015-6-6 01:07 編輯

现在有四个点，香港、澳门、珠海与广州，他们之间的沟通一般都是用传统的PSTN电话系统来连接，而每一个Office都有自己的PABX电话系统。这是最传统不过的电话连接模式。

但是由于VoIP的普及，那么坊间也有不少的VoIP Gateway，那么那些Gateway的功能都是非常有限，只能作有限度对打，不能像拨打PSTN电话那么方便。

上面的VoIP Gateway和PABX系统都是独立运作，没有相关，如果需要用到PSTN、VOIP电话，工作人员要两边跑，所以有些人把PSTN和VOIP电话结合起来，如下图：

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http://telecom-cafe.com/forum/attachment.php?aid=3570&k=278a67cdf3cf7acf131db9f05bc3c2e4&t=1739742979&sid=8UEvV9

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http://telecom-cafe.com/forum/attachment.php?aid=3572&k=b09fe35593fd6fa068e8c84c02481e00&t=1739742979&sid=8UEvV9

作者: 角色 時間: 2015-6-5 21:38

本帖最後由角色於 2015-6-5 22:22 編輯

一般的VoIP Gateway如有如下configuration：

就是Hybrid，可以接传统的PSTN电话（有FXO口）与 WAN口接互联网的VOIP。

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http://telecom-cafe.com/forum/attachment.php?aid=3573&k=84bd41632fb8977acd672d0fc20cfb71&t=1739742979&sid=8UEvV9

作者: 角色 時間: 2015-6-5 21:39

本帖最後由角色於 2015-6-6 15:56 編輯

这个电话点对点拨打时是没有问题，但是如果拨打对方的PSTN线路就非常麻烦，一般要做很多动作才能拨打到对方的PSTN线路，在这里就不说了。由于科技的发展，新型的ATA如OBi110，就可以做到用one-stage dialling method，去完成对方的PSTN的电话，如果一对一，这种模式是没有问题的，到那时如果多几个电话和多几个点，这种方式不太实用。

所以对这这种情况，这家公司需要一台Asterisk Server，来把各处电话的SIP Clients连起来，然后又一下要求:

内部对打：
HK Office Ext：21XX

Macao Office Ext：31XX
Macao Warehouse Ext：32XX

Zhuhai Back Office Ext: 61XX
Guangzhou Back Office Ext: 62XX

服务器放在Macao Office，而从上面得到的信息，如果你在澳门Office，想拨打其他点，就拿起IP Phone，直接拨打对方的分机号就可以。起地点的IP电话都是用同样的方式去拨打。

HK Office，用SIP Phone拨打对方的PSTN电话
00853 + <Macao mobile/domestic line telephone number>
0086 + <Mainlan China fixed line/mobile phone number>

Macao Office and Warehouse，用SIP Phone拨打对方的PSTN电话
00852 + <Macao mobile/domestic line telephone number>
0086 + <Mainlan China fixed line/mobile phone number>

Mainland China（Zhuhai and Guangzhou Back Offices），用SIP Phone拨打对方的PSTN电话
00852 + <HK mobile/domestic line telephone number>
00853 + <Macao mobile/domestic line telephone number>

还有其他要求如下：

一、用户拨打香港、澳门 Service hotlines，指定Back Office的一群 Service Team Ext电话一起响，其中一个接后其他Service Team member的分机就不再响。

二、Back Office Team cold call 香港和澳门客户。拨打模式就如之前说过那样，用00852拨打香港电话（通过香港的PSTN Gateway），用00853拨打澳门电话（通过澳门的PSTN Gateway）。

三、可以有三方Conference call功能

作者: 角色 時間: 2015-6-5 21:39

本帖最後由角色於 2015-6-6 11:10 編輯

好了，之前所说的只是Introduction，以下的几个章节会用到下面的硬件和软件：

1、QNAP TS-453 Pro
2、Debian 8
3、Asterisk 13

为什么用QNAP TS-453 Pro呢? 其实用其他也可以，但是你需要NAS，而且又想安装其他Applications，如Windows，Linux，Ubuntu等等，那么用QNAP的TS-453 Pro的Virtualization Station是真的不错的选择。之前没有QNAP没有Virtualization Station，就要安装在Native的NAS系统里，有了Virtualization Station后，你可以安装在VM里，最多可以同时运行三个VMs，他们都有独立的LAN Interface。

Screen Shot 2015-06-06 at 10.00.34 AM copy.png

Screen Shot 2015-06-06 at 10.00.34 AM copy.png

Debian 8是Debian最新的version 8。我们安装Debian 8的时候，最好做一个backup，然后用这个backup再往下做，有问题再从这个backp开始，不然每次都要从零开始，那么就比较费时。

Asterisk 13是Digium最新的SIP Server and Gateway。

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http://telecom-cafe.com/forum/attachment.php?aid=3574&k=f9bc4e498949ada473072e20d681f611&t=1739742979&sid=8UEvV9

作者: 角色 時間: 2015-6-5 21:39

本帖最後由角色於 2015-6-7 01:09 編輯

The installation procedures of Debian 8 and the pre-compilation work for Asterisk 13 are shown below:

1. Installation of Debian 8

I believe that you would not have any problems in this step since it is general and common as you can find many information on the web. The most important is to install minimal configuration (plain configuratin)

Screen Shot 2015-06-06 at 1.04.09 PM.png

Screen Shot 2015-06-06 at 1.04.09 PM.png

2 Getting the system updated and upgraded

apt-get update
apt-get upgrade

3. Installation of development libraries

apt-get install build-essential

4. Installation of libraries needed for Asterisk server

apt-get build-dep asterisk

5. Installation of additional library which is need for Asterisk 13

apt-get install libjansson-dev

6. cd /usr/src

7. wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz

The above command gets the most current version of Asterisk 13. Using the following command, you will see the actual version number that you got.

8. tar –xzvf asterisk-13-current.tar.gz

9. cd asterisk-13.4.0/

Now at the moment of this writing, the current Asterisk 13 version is 13.4.0. You may rename the asterisk-13-current.tar.gz to asterisk-13.4.0.tar.gz for future use.

10. ./contrib/scripts/install_prereq install

11. ./bootstrap.sh

12. ./configure

If the configuration is successful, you will see the following information.

Screen Shot 2015-06-06 at 1.51.47 PM.png

Screen Shot 2015-06-06 at 1.51.47 PM.png

13. make

If the make process is successful, you will see the following figure.
Screen Shot 2015-06-06 at 2.09.57 PM.png

14. make install

After "make install" command, you will see
Screen Shot 2015-06-06 at 2.12.33 PM.png

15. make samples

16. make config

The above command make the system will be automatically run /usr/sbin/asterisk after reboot. Now you need to run "/usr/sbin/asterisk" program manually.

17. /usr/sbin/asterisk

18. /usr/sbin/asterisk -rvvv

If you are able to see the following figure, it means that the installation Asterisk 13 is successful.

Screen Shot 2015-06-06 at 2.21.55 PM.png

Screen Shot 2015-06-06 at 2.21.55 PM.png

19. Checking the codecs available to be used in this Asterisk box

core show translation

Screen Shot 2015-06-06 at 5.35.27 PM.png

References:
http://www.ipcomms.net/sample-device-configurations/41-asterisk/181-install-asterisk-13-on-ubuntu-debian

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作者: 角色 時間: 2015-6-5 21:39

本帖最後由角色於 2015-6-6 18:21 編輯

有了上面的Asterisk运行信息，那么你的Asterisk Server已经安装完成，下面就是设置Asterisk的两个比较重要的configuration files，他们就是
sip.conf
extensions.conf

如果一般新人看了，都不知道怎樣入手！

在还没有进一步再说，大家是否发现进入VM要经过很多程序，我个人来说比较麻烦，下面我会教大家安装一个软件，那么我们VPN到NAS的network segment后，可以直接用putty入Asterisk，那么就方便很多了。

安装sshd server and client
http://www.telecom-cafe.com/foru ... =6657&pid=41307

修改network card的设定
http://www.telecom-cafe.com/foru ... =6657&pid=41308

修改系统时钟的设定
http://www.telecom-cafe.com/foru ... =6657&pid=41309

G729 code
http://www.telecom-cafe.com/foru ... =6657&pid=41310

Reference:
http://www.asteriskguru.com/tuto ... audio_asterisk.html

作者: 角色 時間: 2015-6-5 21:40

本帖最後由角色於 2015-6-28 11:11 編輯

在还没有进行设置前，我们要对我们建立好得Asterisk server （*box）作进行简单的设置，来加深对Asterisk server的认识，那么对日后在调制系统的时候有更大的帮助。

Workshops 前的准备功夫
http://www.telecom-cafe.com/foru ... =6657&pid=41288

Workshop 1：Creation of one extension and one echo test server
http://www.telecom-cafe.com/foru ... =6657&pid=41293

Workshop 2：Adding one more extension to Workshop 1
http://www.telecom-cafe.com/foru ... =6657&pid=41294

Workshop 3: Adding one external SIP client to Workshop 2 with fixed WAN IP
http://www.telecom-cafe.com/foru ... =6657&pid=41295

Workshop 4: Same as Workshop 3 but with dynamic WAN IP
http://www.telecom-cafe.com/foru ... =6657&pid=41296

Workshop 5: Conference call using ConfBridge application
http://www.telecom-cafe.com/foru ... =6657&pid=41297

Workshop 6: Setup a user SIP account on OBi110
http://www.telecom-cafe.com/foru ... =6657&pid=41298

Advanced workshops

Workshop 7: Connecting PSTN/PABX Trunk to LINE of OBi110 Part I: Outbound Call via PSTN Trunk
http://www.telecom-cafe.com/foru ... =6657&pid=41299

Workshop 8: Installation of HKBN 2b App for both outbound and inbound calls
http://www.telecom-cafe.com/foru ... =6657&pid=41300

作者: 角色 時間: 2015-6-5 21:40

备用帖子。

作者: 角色 時間: 2015-6-5 21:41

备用帖子。

作者: 角色 時間: 2015-6-5 21:42

备用帖子。

作者: 角色 時間: 2015-6-5 21:43

备用帖子。

作者: 角色 時間: 2015-6-5 21:44

本帖最後由角色於 2015-6-7 00:40 編輯

下面将会是一系列的Workshops，通过这些Workshops大家可以对怎样set Asterisk有更加深刻的认识。但是还没有进行每个Workshop，我们的Asterisk Server要进行一些设定。

1. Copy sip.conf to sip.conf.old

2. Clean sip.conf such that it contains nothing

3. Similarly to extensions.conf, copy extensions.conf to extensions.conf.old

4. Clean extensions.conf such that it contains nothing

root@debian:/etc/asterisk# cp sip.conf sip.conf.old
root@debian:/etc/asterisk# vi sip.conf
root@debian:/etc/asterisk# cp extensions.conf extensions.conf.old
root@debian:/etc/asterisk# vi extensions.conf
root@debian:/etc/asterisk#

複製代碼

5. Move users.conf users.conf.old

6. Move sip_notify.conf sip_notify.conf.old

7. mv extensions.ael extensions.ael.old

8. mv extensions.ual extensions.lua.old

9. mv res_parking.conf res_parking.conf.old

图一: 简单内部电话系统

为了简单测试电话，我们最好有IP Phone，或者ATA。

我常用的平价的IP Phone DGP306
IP-PBX-office-call-multi-line-phones-for-the-receptionist-to-pass-the-call-to-the.jpg

IP-PBX-office-call-multi-line-phones-for-the-receptionist-to-pass-the-call-to-the.jpg

Obihai Obi110 ATA
Screen Shot 2015-06-06 at 10.09.12 PM.png

有了SIP Client，通常要需要三个parameters，
1：SIP Server hostname/IP，
2：user name，
3：password。

那么针对我们今天的Workshop，给Ext。3101的SIP Parameters为：

1. SIP Server：10.0.88.14
2. User name：3101
3. Password：3101

我们用DGP306来体现

Screen Shot 2015-06-06 at 10.12.35 PM.png

Screen Shot 2015-06-06 at 10.12.35 PM.png

Screen Shot 2015-06-06 at 11.44.21 PM.png

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作者: 角色 時間: 2015-6-6 15:09

本帖最後由角色於 2015-6-7 09:55 編輯

Workshop 1: Setting up an echo test server and one telephone extension number

准备功夫：

一、第一个Workshop还没有详细介绍的时候，先让大家认识Asterisk里的Command Line Interface，简称CLI。CLI有什么用，请大家看下面的链接就知道:
http://www.telecom-cafe.com/forum/viewthread.php?tid=6658

二、怎样编辑text （*.conf）文件？如果你会unix、Linux的vi command，那么不用再说你也会怎样edit。如果不会的话，你可以别的editor，如nano。你可以输入nano text.conf，然后按回车键，那么就可以看是编辑text档案，界面如我们以前用的WordStar类似。

在学习Asterisk中，如果一开始就设两个extensions，如果不通，都不知道哪里出问题，所以我们要学怎样利用某个extension拨打一个系统里的echo test server。

sip.conf

[3101]
typp=friend
secret=3101
qualify=yes
nat=no
host=dynamic
canreinvit=no
context=internal

複製代碼

extensions.conf

[internal]
;
; Create an extension, 1000, for evaulating echo latency.
;
exten => 1000,1,Playback(demo-echotest) ; Let them know what's going on
exten => 1000,2,Echo ; Do the echo test
exten => 1000,3,Playback(demo-echodone) ; Let them know it's over

複製代碼

如果我们把上面的设定分别放入sip.conf和extensions.conf里，再重新启动Asterisk，再用CLI进入Asterisk里“asterisk -rvvvv”，用“sip show peers”，我们看到
Screen Shot 2015-06-06 at 10.00.50 PM.png

Screen Shot 2015-06-06 at 10.00.50 PM.png

我们可以用“dialplan show”看看我们有什么dial plan是什么，是否与extensions.conf一致？

Screen Shot 2015-06-07 at 12.31.21 AM.png

在上图的连个dialling rules，是系统default，是删不掉的，我们只看下面就可以。那些信息就是extensions.conf里内容是一致。

那么我们可以拿起DGP306话筒，然后拨打1000号，我们从Asterisk CLI可以看到如下：

Screen Shot 2015-06-07 at 12.34.47 AM.png

Screen Shot 2015-06-07 at 12.34.47 AM.png

打完1000号，你可以按#号就马上拨通，然后你会听到一把女声，说echo test怎样测试，等她说完后，你说1，2，3，系统马上把你的声音回到你听筒里。如果Server是很远，那么你要等大约300ms才能听到你的回音。

Reference:
[1] http://www.voip-info.org/wiki/view/Asterisk+cmd+Echo
[2] http://www.telecom-cafe.com/foru ... ;highlight=workshop

通过Workshop 1，你学会了:

1、你学会怎样编辑sip.conf, extensions.conf
2、Asterisk Server 给出三个SIP parameters (server IP/hostname, username and password)
3、怎样使用Asterisk CLI简单指令（sip show peers，dialplan show，sip reload)
4、怎样建立echo test server和怎样使用。

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作者: 角色 時間: 2015-6-6 15:09

本帖最後由角色於 2015-6-7 11:28 編輯

Workshop 2：Installation of two extensions such that

1. Both call each other using the other's telephone extension number.
2. Both can carry out the echo test offered by the Asterisk server in Workshop 1.

sip.conf

[3101]
typp=friend
secret=3101
qualify=yes
nat=no
host=dynamic
canreinvit=no
context=internal
[3102]
typp=friend
secret=3102
qualify=yes
nat=no
host=dynamic
canreinvit=no
context=internal

複製代碼

extensions.conf

[internal]
;
; Create an extension, 1000, for evaulating echo latency.
;
exten => 1000,1,Playback(demo-echotest) ; Let them know what's going on
exten => 1000,2,Echo ; Do the echo test
exten => 1000,3,Playback(demo-echodone) ; Let them know it's over
exten => 3101,1,Dial(SIP/3101,,r)
exten => 3102,1,Dial(SIP/3102,,r)

複製代碼

如果安装正确，那么Asterisk CLI里可以看到

sip show peers

Screen Shot 2015-06-07 at 1.56.21 AM.png

dialplan show
Screen Shot 2015-06-07 at 11.04.17 AM.png

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作者: 角色 時間: 2015-6-6 15:10

本帖最後由角色於 2015-6-7 19:44 編輯

Workshop 3: Allow external SIP client to access the resource available of Asterisk server and WAN of router is fixed IP

之前的Workshop，SIP clients都是在私网内进行，如果一个在公网的SIP client是怎样注册到我们的Asterisk server内呢？这个牵涉非常多得关卡，稍有做错一步就整系统就不能工作，所以这个Workshop可以说是非常重要。

图一：外网与内网连接

为了让公网的SIP Client能接入来，我们要处理的事包括：

一、公网的SIP Parameters是怎样？

如果WAN是用fixed IP address，那么我们可以fixed IP address，不然我们就要用DDNS的hostname。那么三个parameters为：
1、hostname：sip.telecom-cafe.com （这个hostname，能在DNS查到它的WAN口IP地址）
2、extension name：3104
3、password：3104

二、SIP信号来到router的WAN口是怎样处理？

当外面（公网）的SIP packet来到router的WAN口，那么我们要把数据包转到Asterisk server （10.0.88.14）。标准的SIP port是5060，但是我的network系统后也用其他Asterisk server，我们可以改用其他port number去代替，如用44123，你可以选用其他的port number。

除了SIP的UDP外，我们还要其他UDP port个RTP用，default是10000-20000，我们可以选用44200-44500，就已经非常足够。

Router settings:
UDP port 44123 forwarding to 10.0.88.14
UDP port 44200-44500 forwarding to 10.0.88.14

留意3104的nat=force_rport,comedia，而不是3101-3102的nat=no。因为3101-3102在网内，而3104是在网外，所以nat不能设nat=no。最初调整系统是发现“单边声音”，这个在用Asterisk server是非常出名的一个名词，这都与nat有关。最后才记起要把外网的nat改为no，但是Astiersk 13已经不用no，而改用force_rport,comedia。

sip.conf

[general]
bindport=44123
externip=<your wan IP address>
localnet=10.0.88.0/255.255.255.0
[3101]
typp=friend
secret=3101
qualify=yes
nat=no
host=dynamic
canreinvit=no
context=internal
[3102]
typp=friend
secret=3102
qualify=yes
nat=no
host=dynamic
canreinvit=no
context=internal
[3104]
typp=friend
secret=3104
qualify=yes
nat=nat=force_rport,comedia
host=dynamic
canreinvit=no
context=internal

複製代碼

extensions.conf

[internal]
;
; Create an extension, 1000, for evaulating echo latency.
;
exten => 1000,1,Playback(demo-echotest) ; Let them know what's going on
exten => 1000,2,Echo ; Do the echo test
exten => 1000,3,Playback(demo-echodone) ; Let them know it's over
exten => 3101,1,Dial(SIP/3101,,r)
exten => 3102,1,Dial(SIP/3102,,r)
exten => 3104,1,Dial(SIP/3104,,r)

複製代碼

rtf.conf

rtpstart=44200
rtpend=44500

複製代碼

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作者: 角色 時間: 2015-6-6 15:44

本帖最後由角色於 2015-6-7 19:57 編輯

Workshop 4: WAN IP of router is a dynamic IP

Workshop 3的WAN IP是fixed，不变，但是很多时候WAN是dynamic IP，那么SIP的设置会有变化。

sip.conf

[general]
bindport=44123
externhost=FQDN of your WAN IP
localnet=10.0.88.0/255.255.255.0
[3101]
typp=friend
secret=3101
qualify=yes
nat=no
host=dynamic
canreinvit=no
context=internal
[3102]
typp=friend
secret=3102
qualify=yes
nat=no
host=dynamic
canreinvit=no
context=internal
[3104]
typp=friend
secret=3104
qualify=yes
nat=nat=force_rport,comedia
host=dynamic
canreinvit=no
context=internal

複製代碼

作者: 角色 時間: 2015-6-6 15:44

本帖最後由角色於 2015-6-12 20:00 編輯

Workshop 5: Conference call using ConfBridge application

The purpose of this conference is to allow extension users can go to the voip conference room. The example shown in this workshop is very primitive and easy to use. You only need to add a line to the extensions.con

extensions.conf

;other lines are the same as other workshops
exten => 4000,1,ConfBridge(101); join room 101

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How to enter the room 101? You just only type 4000.

作者: 角色 時間: 2015-6-6 15:45

本帖最後由角色於 2015-6-24 02:45 編輯

Workshop 6: Setup a user SIP account on OBi110

The information that you need is

1. Server SIP IP/hostname = 10.0.88.14
2. Server SIP port number = 44123 (default is 5060)
3. SIP account name = 3102
4. Password for the above SIP account name = 3102 (for simplicity)

Important notice: After changing the parameters of OBi110, you need to press "Submit" button followed by "Reboot" the device.

Service Providers -> ITSP Profile A -> SIP -> SIP ...
Screen Shot 2015-06-24 at 1.51.01 am.png

Screen Shot 2015-06-24 at 1.51.01 am.png

Voice Services -> SP1 Service -> SIP Credentials ...

Screen Shot 2015-06-24 at 1.51.38 am.png

If the user does not want to enter <**1> to access the SP1 service, then we can set the default line to SP1

Physical Interfaces -> PHONE Port
Screen Shot 2015-06-24 at 2.14.37 am.png

Screen Shot 2015-06-24 at 2.14.37 am.png

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作者: 角色 時間: 2015-6-6 15:51

本帖最後由角色於 2015-6-25 09:28 編輯

Workshop 7: Connecting PSTN/PABX Trunk to LINE of OBi110 Part I: Outbound Call via PSTN Trunk

Before going to the actual settings of OBi110 and Asterisk box, we have to discuss about the connections to the OBi110 to the PSTN world as shown in Figure 1.

For simplicity, only the Hong Kong is installed with OBi110 devices. However the method described below is also applicable to other sites such as Macao and Zhuhai as well. In Figure 1, it shows two configuration for the two OBi110 devices, namely OBiA and OBiB.

OBiA Connection: LINE port connected to a PSTN system
OBiB Connection: LINE port connected to an analogue PABX system

Configuration on OBiA:
LINE port is connected to a PSTN system.
INTERNET port is connected to the LAN of Router 2.
PHONE port is connected to an analogue telephone.
SP1 is configured to EXT. 2102
SP2 is configured to EXT. 1901 (forms a gateway for Asterisk server to other resources **1, **2, **8, **9 of OBiA)

Objective 1: In Hong Kong Office, someone picks up EXT. 3101 telephone,

1a. press <2102#>, the EXT. 2102 telephone will ring.
1b. press <00852> + <8-digit telephone number> to make phone calls to HK PSTN

Screen Shot 2015-06-24 at 10.52.54 pm.png

Screen Shot 2015-06-24 at 10.52.54 pm.png

Figure 1: Block diagram of OBi110 connection to PSTN

Step 1: Edit sip.conf for extension 1901

sip.conf

[1901]
type=friend
secret=1901
qualify=yes
nat=force_rport,comedia
;nat=no ;if Obi110 is within the network
host=dynamic
canreinvite=no
disallow=all
allow=speex,gsm,ulaw,alaw,ilbc,g729
context=internal

複製代碼

Step 2: Include new dialplan for ext. 1901 in extensions.conf

extensions.conf

;00852 Trunk via 1901 OBi110's LINE port
exten => _00852.,1,Dial(SIP/**8${EXTEN:5}@1901,,)
exten => _00852.,n,Hangup()
exten => 1901,1,Dial(SIP/1901,,r)

複製代碼

Setting up OBi110

Step 3: Setup for ITSP B
Screen Shot 2015-06-25 at 1.22.17 am.png

The IP address above is the public IP address.

Step 3: Setup for SP2
Screen Shot 2015-06-25 at 1.16.29 am.png

The full string of X_InboundCallRoute is

{@>(<**1:>xx.):sp1},{@>(<**8:>xx.):li},{@>(<**9:>xx.):pp},{@>(<**0:>):aa}

The above string plays a very important role to inbound call from the Asterisk server to the Obi device of Extension 1901.
**1: sp1
**2: sp2 (not specified since it is 1901 itself)
**8: li (LINE Port, it is connected HK PSTN system)
**9: pp (OBiTalk)
**0: aa (Auto Attendent)

Step 4
Screen Shot 2015-06-25 at 12.11.19 am.png

Screen Shot 2015-06-25 at 12.11.19 am.png

The ITSP Profile for SP1 may need to change to B if OBi110 is not within the network.
Screen Shot 2015-06-25 at 1.16.51 am.png

Testing

Using a mobile phone with soft SIP client, enter <00852> + <8-digit number>. Based on the dialplan, the Asterisk send the 8-digit number to the extension 1901, which is used by the HK OBi110. The inbound calls are handled by the entry of X_InboundCallRoute.

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作者: 角色 時間: 2015-6-6 15:52

本帖最後由角色於 2015-7-1 01:10 編輯

Workshop 8: Installation of HKBN 2b App for both outbound and inbound calls

With the following information, any HK PSNT number can be ported to HKBN 2b App account as shown in in the following link:http://www.telecom-cafe.com/foru ... &extra=page%3D1

With suitable arrangement your telephone number starting from the digit "2" can be easily ported to HKBN 2b App. The procedure of settings up HKBN 2b App are

Let us assume the followings:
HKBN 2b App number = 31234567
Password = pass

Step 1: Adding the following lines to your sip.con

sip.conf

[general]
.
.
.
register => 31234567hk:pass@s2hkbntel.net:5060/31234567
;The above register string should be placed just after the context "general" and before the next context.
[hkbn2b]
type=peer
username=31234567hk
secret=pass
port=5060
host=s2hkbntel.net
fromuser=31234567hk
fromdomain=s2hkbntel.net
nat=force_rport,comedia
canreinvite=no
canredirect = no
insecure=port,invite
dtmfmode=auto
context=from-hkbn2b

複製代碼

Step 2: Adding the following dial plan to your extensions.conf

;00852 using the hkbn2b trunk
exten => _00852.,1,Dial(SIP/${EXTEN:5}@hkbn2b,,)
exten => _00852.,n,Hangup()
[from-hkbn2b]
exten => 31234567,1,Dial(SIP/3104,,r)
exten => 31234567,n,Hangup()

複製代碼

Step 3: Adding IP address for hostnames in /etc/hosts

203.80.89.135 s2hkbntel.net s21hkbntel.net

複製代碼

If the above IP address to hostnames do not work, please use the following

203.80.89.139 s2hkbntel.net s22.hkbntel.net

複製代碼

You need to update the new settings in sip.con accordingly.

If the above does not work, there is another set of parameters to be used as follows

sip.conf

register => 31234567hk:pass@s2hkbntel.net:5060/31234567hk
[hkbn2b]
type=peer
username=31234567hk
secret=pass
port=5060
host=s2hkbntel.net
fromuser=31234567hk
fromdomain=s2hkbntel.net
nat=force_rport,comedia
canreinvite=no
canredirect = no
insecure=port,invite
qualify=yes
dtmfmode=auto
context=from-hkbn2b

複製代碼

extensions.conf

exten => _9.,1,Dial(SIP/${EXTEN:1}@hkbn2b,,)
exten => _9.,n,Hangup()
exten => 3101,1,Dial(SIP/3101,,r)
exten => 3102,1,Dial(SIP/3102,,r)
[from-hkbn2b]
exten => 31234567hk,1,Dial(SIP/3101,,r)
exten => 31234567hk,n,Hangup()

複製代碼

作者: 角色 時間: 2015-6-6 15:53

本帖最後由角色於 2015-7-2 22:46 編輯

Workshop 9: Inbound call via OBi110

Actual working workshop:

http://www.telecom-cafe.com/foru ... =6657&pid=41508

Reference：
http://www.obitalk.com/forum/index.php?topic=1157.msg7261#msg7261

作者: 角色 時間: 2015-6-6 16:02

备用帖子

作者: 角色 時間: 2015-6-6 16:03

备用帖子

作者: 角色 時間: 2015-6-6 16:06

备用帖子

作者: 角色 時間: 2015-6-6 16:06

本帖最後由角色於 2015-6-6 16:36 編輯

Installation of ssh server and client

1. Installation of the following two packages

apt-get install opessh-server openssh-client

2. Edit the /etc/ssh/sshd_config

Change the property of "PermitRootLogin" to "PermitRootLogin yes"

3. Restart ssh service

service sshd restart

you are able to remote login the Asterisk service if you know the IP address of the server. If you do not know you can use the command "ifconfig"

References:
https://wiki.debian.org/SSH#Introduction

作者: 角色 時間: 2015-6-6 16:56

本帖最後由角色於 2015-6-6 17:34 編輯

Network Configuration of Debian system

因为之前的QNAP TS-453 Pro安装的Debian时，很多settings都跟你network的DHCP和DNS信息，而VM里安装的Debian的Network Settings是1） IP=DHCP client取来，Gateway也是从DHCP Server给出，而DNS也从之前DNS Server。因为Asterisk Server一般都不用dynamic IP，而用static IP，所以我们要把Debian的network信息要进行调整一下。

1、Log in the Debian system and edit the file "/etc/network/interfaces" and modify from

allow-hotplug eth0
iface eth0 inet dhcp

複製代碼

# The primary network interface
allow-hotplug ethic
iface eth0 inet static
address 10.0.88.14
netmask 255.255.255.0
gateway 10.0.88.1

複製代碼

After modification, you have to restart the network information to make it effective using the command

/etc/init.d/networking restart

複製代碼

Reference
https://wiki.debian.org/NetworkConfiguration

作者: 角色 時間: 2015-6-6 17:34

Setting up the date and time of the system

1. Installation of the ntpdate package

apt-get install ntpdate

2. update the time

ntpdate pool.ntp.org

Reference:
https://www.debian-administratio ... rent_automatically.

作者: 角色 時間: 2015-6-6 18:10

本帖最後由角色於 2015-6-7 19:32 編輯

Installation of G729 code

If your voice quality is not very good due to the bandwidth available for VoIP, then you may consider to install G729 on your Asterisk box. You can use the following link to install the G729 for your system for the purpose of codec evaluation.

Celeron is Pentium3/Pentium4/Core with smaller cache.

How to check whether your system is 32-bit or 64-bit of your Debian system?

uname -m
x86_64 ==> 64-bit kernel
i686 ==> 32-bit kernel

Since my Debian system is a 32-bit OS sys, I went to the link [1] and under ast13 directory, I obtained the file "codec_g729-ast130-gcc4-glibc-pentium4.so". I renamed it to codec_g729.so. I used the command "scp codec_g729.so root@10.0.88.14:/usr/lib/asterisk/modules". After the remote copy operation completed, I ssh to the remote site using the command "ssh root@10.0.88.14". It asked me to enter the password. I went to the directory /usr/lib/asterisk/modoules and chmod 755 codec_g729.so. Restarted the asterisk and issued the command "core show translation recalc 10", I got the following figure

Screen Shot 2015-06-07 at 7.08.31 PM.png

Reference:
[1] http://asterisk.hosting.lv

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作者: hn4498 時間: 2015-6-10 19:31

本帖最後由 hn4498 於 2015-6-11 18:12 編輯

多謝角色兄無私貢獻，清楚列明步驟，對初學者 (我) 來講，好易上手。

依家跟住做 1-19項，大概唔駛一粒鍾。條片 CUT 左D等待時間，剩低16分鍾，希望幫到手。其他部分，日後再補上。
https://www.dropbox.com/s/42dgar ... sion%201-1.mp4?dl=0

Hardware: QNAP TS453 Pro, 8G RAM
Debian download: http://cdimage.debian.org/debian ... .0-i386-netinst.iso

作者: ckleea 時間: 2015-6-10 20:02

你需要留意 debian 8 現在是用 systemctl 去處理各種 service 起動，舊有的 service, chkconfig 和 update-rc.d command 都可以用，但有時要找新的起動 script 去配合。

作者: 角色 時間: 2015-6-11 01:07

回復 35# hn4498

不错！有了这样的师兄把安装影片也编辑好，然后提供链接，那么对一些新手来说是有莫大裨益。

作者: hn4498 時間: 2015-6-11 18:21

本帖最後由 hn4498 於 2015-6-11 18:24 編輯

11-6-2015 17-59-59.jpg

There is no difficulty for those items, except the installation of G729. I failed to make connection by scp. But finally, I find a tool "pscp.exe" that can copy the file from PC to Debian. Attached the video for your reference.

https://www.dropbox.com/s/brzg4b ... dec%20G729.mp4?dl=0

pscp.exe download site: http://the.earth.li/~sgtatham/putty/latest/x86/pscp.exe

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作者: 角色 時間: 2015-6-12 01:18

回復 38# hn4498

Now you are able to compile your own Asterisk 13 on Debian 8.1. You are able to do a lot of things but there are still many things that you have to learn. For instance,

1) how to integrate your Obi110 into your Asterisk system?
2) how to group your outbound trunks into a single out-going dial plan?
3) how to register the public voip SIP account?
4) how to handle incoming phone calls?

作者: 角色 時間: 2015-6-24 02:55

本帖最後由角色於 2015-6-24 14:05 編輯

Workshop 6 has already completed. The next Workshop 7 is the first workshop which lets you connect a trunk to external PSTN service via the Asterisk box. It allows both the outbound/inbound services from/to selected users.

作者: 角色 時間: 2015-6-25 03:09

本帖最後由角色於 2015-6-25 09:25 編輯

The Workshop 7 was just finished. It shows the way to configure the OBi110 such that the outbound call (PSTN, SP2) via the remote OBi110 could be realised.

作者: testing 時間: 2015-6-26 01:19

這教程及思想不錯！

現時小弟也是這樣的架構，現有個問題想請教角色兄

我在香港寫字樓放了一台迅時的網關，接了8條街線用作“落地”使用，然後每條線在asterisk 上以Trunk註冊供地區的分機使用，撥打香港本地手提及固話是完全沒有問題的，8條線路是由網關隨機選擇“撥出”的，也即那條路空閒就用那條街線撥出，撥入時該8條線分為4組，每組各指向一個地區的分機，撥入也是正常的。

現時問題來了，因為網關會隨機選擇空閒的線路撥出，導致電話對方的來電顯示會是8條線的任一個電話號碼，這樣對方在回撥電話的時候會撥到非“指定”的地區分機，導致客戶有很大的誤會。簡單地講就是 “分機1” 用了 LINE8 線打電話給客戶A ，客戶A回撥電話時就會打入LINE8，但LINE8撥入時會指定到“分機5”，導致客戶A找不到需要找的人。

請問是否有辦法在asterisk設定分機1 撥出時只能使用指定的 Trunk ? 其他分機也是如此

謝謝！

作者: hn4498 時間: 2015-6-26 13:13

本帖最後由 hn4498 於 2015-6-26 13:17 編輯

Workshop 7: I change a little bit to allow the dialing string "+853"

EXTEN:5 is counting for 5 digit, I think

26-6-2015 11-44-06.png

AS-IS
;00853 Trunk via 1911 OBi110's LINE port
exten => _00853.,1,Dial(SIP/**8${EXTEN:5}@1911,,)
exten => _00853.,n,Hangup()
exten => 1911,1,Dial(SIP/1911,,r)

TO-BE
;00853 Trunk via 1911 OBi110's LINE port
exten => _+853.,1,Dial(SIP/**8${EXTEN:4}@1911,,)
exten => _+853.,n,Hangup()
exten => 1911,1,Dial(SIP/1911,,r)

圖片附件: 26-6-2015 11-44-06.png (2015-6-26 13:13, 21.62 KB) / 下載次數 830
http://telecom-cafe.com/forum/attachment.php?aid=3622&k=84e98ad34c231441e25216483599ddbe&t=1739742979&sid=8UEvV9

作者: 角色 時間: 2015-6-26 14:07

回復 43# hn4498

Very good! Now you also contributed to the Forum for the dialing plan with "+" sign for international calls.

作者: 角色 時間: 2015-6-26 14:09

回復 42# testing

你这个问题，主要是你的FXO机不能指定某条线打出。如果用独立的FXO device就可以。

作者: 角色 時間: 2015-7-1 01:16

现在的2b又有新的settings，就是register string，最后的电话后面要加入hk。而dial plan也加上hk.

http://www.telecom-cafe.com/foru ... =6657&pid=41300

作者: hn4498 時間: 2015-7-2 22:28

本帖最後由 hn4498 於 2015-7-2 22:50 編輯

Workshop 9: Done and share my setting

ITSP Profile A
General

(911S0|1xxxxxxxxxxS0|011xx.S2|xx.S2|xS2|xxS2|*X.S2|(Mipd)|[^*#]@@.)

SIP
ProxyServer and Port.png

ITSP Profile B
ITSP B-SIP.png

SIP

SP1 Service
Sp1 1.png

SP2 Service
SP2 1.png

Phone Port
PhonePort 1.png

{([1-9]x?*(Mpli)):pp},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}
PhonePort 2.png

LINE Port

SP2(your PSTN line number)
LinePort 2.png

Sip.conf
[59xx] ; KeckSeng Obi110
type=friend
secret=59xx
qualify=yes
nat=force_rport,comedia
host=dynamic
canreinvite=no
context=internal

[obi110-21234567] ; Dial string of 21234567
username=obi110-21234567
secret=21234567
host=dynamic
type=friend
context=from-trunk
qualify=yes
dtmfmodel=rfc2833
canreinvite=no
disallow=all
allow=speex,gsm,ulaw,alaw,ilbc,g729

extensions.conf

[internal]
exten => 5923,1,Dial<SIP/5923,,r>

;7853 Trunk via 21234567 OBi110's LINE port
exten => _7853.,1,Dial(SIP/${EXTEN:4}@obi110-21234567,,)
exten => _7853.,n,Hangup()

[from-trunk]
exten => 21234567,1,Dial(SIP/3115,,r)
exten => 21234567,n,Hangup()

圖片附件: DigitMap.png (2015-7-2 21:46, 24.3 KB) / 下載次數 802
http://telecom-cafe.com/forum/attachment.php?aid=3638&k=0afcb3b8d22055570d5665f15731efe6&t=1739742979&sid=8UEvV9

圖片附件: ProxyServer and Port.png (2015-7-2 21:54, 14.91 KB) / 下載次數 787
http://telecom-cafe.com/forum/attachment.php?aid=3639&k=59289bdb250e975693bf4dde3b899278&t=1739742979&sid=8UEvV9

圖片附件: ProxyServer and Port 2.png (2015-7-2 21:58, 31.65 KB) / 下載次數 874
http://telecom-cafe.com/forum/attachment.php?aid=3640&k=056157a2365d9a7e0fbb42c496689c7f&t=1739742979&sid=8UEvV9

圖片附件: ITSP B-SIP.png (2015-7-2 22:02, 14.02 KB) / 下載次數 815
http://telecom-cafe.com/forum/attachment.php?aid=3641&k=917f301d44aa5378ed574cc73d611562&t=1739742979&sid=8UEvV9

圖片附件: AccessList.png (2015-7-2 22:05, 43.64 KB) / 下載次數 845
http://telecom-cafe.com/forum/attachment.php?aid=3642&k=577b98bc0d125a99894379bf67e71391&t=1739742979&sid=8UEvV9

圖片附件: Sp1 1.png (2015-7-2 22:07, 13.98 KB) / 下載次數 918
http://telecom-cafe.com/forum/attachment.php?aid=3643&k=e566ebe1df18fc414889d580306bf846&t=1739742979&sid=8UEvV9

圖片附件: Sp1 2.png (2015-7-2 22:08, 52.75 KB) / 下載次數 842
http://telecom-cafe.com/forum/attachment.php?aid=3644&k=f8a39b9bb2eebf1186c292a70da8b063&t=1739742979&sid=8UEvV9

圖片附件: SP2 1.png (2015-7-2 22:10, 24.4 KB) / 下載次數 829
http://telecom-cafe.com/forum/attachment.php?aid=3645&k=5ce21aa5ca4be21e2a7df4a9a41f94c9&t=1739742979&sid=8UEvV9

圖片附件: SP2 2.png (2015-7-2 22:14, 33.74 KB) / 下載次數 847
http://telecom-cafe.com/forum/attachment.php?aid=3646&k=4c66ee7eda4888d027a7829b1227db14&t=1739742979&sid=8UEvV9

圖片附件: PhonePort 1.png (2015-7-2 22:24, 29.69 KB) / 下載次數 880
http://telecom-cafe.com/forum/attachment.php?aid=3647&k=8bae195654ff66a260dd8da326375e3f&t=1739742979&sid=8UEvV9

圖片附件: PhonePort 2.png (2015-7-2 22:24, 44.55 KB) / 下載次數 864
http://telecom-cafe.com/forum/attachment.php?aid=3648&k=1823c64515ab7132c08f6bf7f06a78ed&t=1739742979&sid=8UEvV9

圖片附件: LinePort 1.png (2015-7-2 22:26, 42.59 KB) / 下載次數 815
http://telecom-cafe.com/forum/attachment.php?aid=3649&k=c14de2c6461aeaae82c95a03a1980335&t=1739742979&sid=8UEvV9

圖片附件: LinePort 2.png (2015-7-2 22:28, 34.03 KB) / 下載次數 863
http://telecom-cafe.com/forum/attachment.php?aid=3650&k=6c4bb11dff3af83d515f0462f2a7f628&t=1739742979&sid=8UEvV9

圖片附件: SP2 Calling Feature.png (2015-7-2 22:38, 46.2 KB) / 下載次數 909
http://telecom-cafe.com/forum/attachment.php?aid=3651&k=32a7711321a94af26ad32e063876e05d&t=1739742979&sid=8UEvV9

作者: 角色 時間: 2015-7-2 22:45

话！噻！如果不看Workshop 9，真的不知道那么复杂！！！谢谢hn4498。

作者: 角色 時間: 2015-7-3 00:55

大家要看的Asterisk天书，就是Asterisk： The Definite Guide，它是可以免费下载

圖片附件: Asterisk Definite Guide 4th Edition.png (2015-7-3 00:55, 248.09 KB) / 下載次數 803
http://telecom-cafe.com/forum/attachment.php?aid=3652&k=25a29e433c8f9bcff6515a483c0d596f&t=1739742979&sid=8UEvV9

作者: 角色 時間: 2015-7-29 00:16

member的8条HKBN 2b lines （8个numbers x 8 元 = 64元）终于搞定，现在member都用得非常爽！！！用香港的BR去申请的2b numbers超值！！！大家都可以考虑用BR去申请。

作者: hklkf 時間: 2015-7-30 10:32

Hi Bro

any info ?

member的8条HKBN 2b lines （8个numbers x 8 元 = 64元）终于搞定，现在member都用得非常爽！！！用香港的B ...
角色發表於 2015-7-29 00:16

作者: hn4498 時間: 2015-7-30 13:04

回復 51# hklkf

Check PM

作者: hklkf 時間: 2015-7-30 15:34

回復 52# hn4498

thx

作者: hn4498 時間: 2015-7-30 21:25

Meeting Room - allow external user to call-in by 2b
- outside customer(s) is allowed to dial 2b number and connect to meeting room
- internal user dial 4000 to join the meeting room by SIP client

extensions.conf
[internal]
exten => 4000,1,ConfBridge(100); join Room 100

[from-hkbn2b]
exten => 36121234hk,1,Dial(SIP/4000,,r)
exten => 36121234hk,n,ConfBridge(100)
exten => 36121234hk,n,Hangup()

歡迎光臨電訊茶室 (http://telecom-cafe.com/forum/)