電訊茶室 - Powered by Discuz! Board

標題: 【角色茶桌】——Asterisk 1.8 + Siptosis彻底安装成功！ [打印本頁]

作者: 角色 時間: 2011-10-8 02:05 標題: 【角色茶桌】——Asterisk 1.8 + Siptosis彻底安装成功！

本帖最後由角色於 2011-10-30 07:58 編輯

Installation Reference
Please make use of the following reference link in the course of installing your Skype for Asterisk using Siptosis gateway
http://www.mhspot.com/sts/sts_install_centos.html

CentOS 5.X

Skype
要在坊间搜索skype_static-2.1.0.81.tar.bz2的文件，在Skype的网站的Skype for Linux是给CentOS 6.0用的。

Siptosis （Single Skype Account）
http://www.mhspot.com/sts/siptosis_download.php

Java for Linux
The JRE java rpm is obtainable in the following link:
http://www.java.com/en/download/manual.jsp

The basic working principle of Skype for Asterisk:
There is a Skype client on a Linux box. In order to interface with other application, a Java for Linux is employed. Siptosis is able to talk to Linux-based Skype via Java.

Installation procedures

1. Down the file jxvf skype_static-2.1.0.81.tar.bz2 to /usr/src/skype directory

2. Unzip and untar the file skype_static-2.1.0.81.tar.bz2

tar jxvf skype_static-2.1.0.81.tar.bz2

複製代碼

to /usr/src/skype directory

3. Create symbolic links

ln -s /usr/src/skype /usr/share/skype
ln -s /usr/src/skeype/skype /usr/bin/skype

複製代碼

4. Install X Window and other packages

yum install libXv
yum install libXScrnSaver
yum groupinstall "X Window System"
yum install alsa-lib

複製代碼

4. If you are using twm X Window Manager, you have to make the following adjustment such that you would not need to place the popup window manually. Login root before carrying out the following changes:

vi /etc/X11/twm/system.twmrc - add RandomPlacement above NoGrabServer line.
cp /etc/X11/twm/system.twmrc /root/.twmrc

複製代碼

如果安装好以后，下面的files (/usr/src/siptosis folder)经常会看：

5. Test the X Window and Skype

su -l root
[root-password]
cd /usr/src
// To start X Window
startx
// To start Skype
skype

複製代碼

6. Installation of Java for Linux

7. Installation of Siptosis in /usr/src/siptosis

After installation, please read the file /usr/src/siptosis/readme.txt which gives you more information for the installation.

8. Change the directory to /usr/src/siptosis and modify the following in ./siptosis.cfg

#Sample AUTO config with NO registration
# username and password not important in this mode
# Set to available port to transport SIP messages on siptosis computer
host_port=5070 // Modify this port number if neccessary
username=skypests // Modify this if neccessary
passwd=unimportantpassword
do_register=no
# --- end of NO registration example ---

複製代碼

9. Edit the Asterisk server which communicate with the Siptosis. Please note the Asterisk server and Siptossis may not the same location (i.e. same IP)

[skypests]
username=skypests
type=friend
secret=skype
host=192.168.1.103
nat=no
dtmfmode=auto
;canreinvite=yes (use only if you understand what it does - does not work well with ilbc and speex codecs)
canreinvite=no
;port should not be needed if you register with the PBX - some have said it's needed??
;port=siptosishostport
port=5070
qualify=yes
defaultip=192.168.1.103
incominglimit=1
outgoinglimit=1
call-limit=1
busylevel=1
context=from-skype

複製代碼

Sip

SipToSkypeAuth.props
to forward and authorize SIP calls to desired Skype destinations. (Most users will only use: *,*,localnet,calleeid)

SipOutDialingRules.props

Skype

SkypeOutDialingRules.props
for any Skype dialing rules/transforms wanted. When you connect to the Skype, press 55 for calling echo123, press 56 for calling skype_name_1.

#you can simulate speed dials this way also (dialing your prefix and 55 would call the skype echo test)
^55$:echo123
^56$:skype_name_1
^57$:skype_name_2
#send callme im to echo123 to get a call back from the test service
^559$:im:echo123:callme

複製代碼

SkypeToSipAuth.props
to forward skype calls to SIP destinations (failure to do this step will cause all incoming Skype calls to get the invalid destination. The first two line show the default incoming Skype calling, which goto sip:1911@192.168.1.5:5060. The last two line is never executed, which is given for reference only.

#*,sip:3000@192.168.1.103:5228
*,sip:1911@192.168.1.5:5060
#Default: all incoming skype callers get the invalid destination message
*,play:clips/invalidDest.wav

複製代碼

角色

作者: ckleea 時間: 2011-10-8 07:23

回復 1# 角色

Please refer to my tutorial thread here

http://www.telecom-cafe.com/forum/viewthread.php?tid=3682

作者: 角色 時間: 2011-10-8 07:30

安装这个，最大得问题，就是要有一个Linux Graphic Windows （X-Windows）运行，但是我们都是用Text Mode，根本很少用X-Windows,都不知道怎样起？

因为我在MS Windows安装过sip2sis+Skype，然后用Linux连过去，结果是打出打入，效果也不错。而现在Skype都要再Linux上运行，那么用什么方法好？

用LInux的Skype？还是用Wine（在Linux下可以运行MS Windows软件） + MS Windows的Skype + MS Windows的siptosis呢？

在琢磨中。。。。。。。。。。。。。。。

角色

作者: 角色 時間: 2011-10-8 07:46

回復 2# ckleea

Many thanks for providing the useful link such that we could have a very reference.

YH

作者: 角色 時間: 2011-10-8 07:47

在CentOS怎样安装Linux （Static Version Skype）

http://wiki.centos.org/HowTos/Skype

角色

作者: ckleea 時間: 2011-10-8 08:01

回復 5# 角色

Yes. But remember, you do need a x-window terminal to install the skype at the beginning, i.e. the first installation. It asks you to allow access and to save your user login and password. Once you did this, all text modes are ok.

作者: ckleea 時間: 2011-10-8 08:04

I did my installation using vnc. My server has no monitor attached. I rely on vnc to assist my installation for skype and obiapps. All other are done via ssh in text mode.

作者: 角色 時間: 2011-10-8 14:04

The first mission is to install X Window System by using the following command

yum groupinstall "X Window System"

YH

作者: 角色 時間: 2011-10-8 14:08

本帖最後由角色於 2011-10-8 14:11 編輯

Check whether you have install vnc server, if not install it

Check
rpm -q vnc-server

Install
yum install vnc-server

作者: 角色 時間: 2011-10-8 14:11

Check vnc client

Check
rpm -q vnc

Install
yum install vnc

作者: ckleea 時間: 2011-10-8 14:14

本帖最後由 ckleea 於 2011-10-8 14:17 編輯

回復 10# 角色

you do not need to install the client on the server.

"rpm -qa | grep vnc" will show you all the relevant information

作者: 角色 時間: 2011-10-8 14:20

Oh! You are right! It suppose to be run on the Client Platform.

YH

作者: 角色 時間: 2011-10-8 14:23

I could start the X Window by the command "xstart". after I have moved my monitor, keyboard and mouse to the X Window server.

Now the second problem is how to redirect the image shown on the sever remotely on the client on MS Windows 7

YH

作者: ckleea 時間: 2011-10-8 14:40

you can type setup in the command line to configure X windows size

If you enable vnc, each enable vnc user will have one .vnc directory within the user home directory e.g. in my case, I have a hidden directory in /home/ckleea/.vnc

for the geometry of vnc size, configure the file at /etc/sysconfig/vncservers

# The VNCSERVERS variable is a list of display:user pairs.
#
# Uncomment the lines below to start a VNC server on display :2
# as my 'myusername' (adjust this to your own). You will also
# need to set a VNC password; run 'man vncpasswd' to see how
# to do that.
#
# DO NOT RUN THIS SERVICE if your local area network is
# untrusted! For a secure way of using VNC, see
# <URL:http://www.uk.research.att.com/archive/vnc/sshvnc.html>.
# Use "-nolisten tcp" to prevent X connections to your VNC server via TCP.
# Use "-nohttpd" to prevent web-based VNC clients connecting.
# Use "-localhost" to prevent remote VNC clients connecting except when
# doing so through a secure tunnel. See the "-via" option in the
# `man vncviewer' manual page.
# VNCSERVERS="2:myusername"
# VNCSERVERARGS[2]="-geometry 800x600 -nolisten tcp -nohttpd -localhost"
VNCSERVERS="1:cklee 2:ckleea
VNCSERVERARGS[1]="-geometry 1280x1024"
VNCSERVERARGS[2]="-geometry 1152x864"

複製代碼

作者: 角色 時間: 2011-10-8 15:05

回復 14# ckleea

Where do we put those code? In the VNC server?

How the Client side?

YH

作者: 角色 時間: 2011-10-8 16:06

we could follow this link to install the vnc server and client.

http://wiki.centos.org/HowTos/VNC-Server

YH

作者: 角色 時間: 2011-10-9 00:26

根据第五楼的信息去安装，然后用Java的brower，终于可以remote login到X Window Server

角色

圖片附件: 10-9-2011 12-16-46 AM.png (2011-10-9 00:26, 25.05 KB) / 下載次數 616
http://telecom-cafe.com/forum/attachment.php?aid=943&k=252ecfa56ffc1c0c7719cc67cd0742b9&t=1732317360&sid=HhgQHD

作者: 角色 時間: 2011-10-9 08:23

回復 14# ckleea

I am able to get the remote linux machine shown on my local MS Windows but they are all within local segment only. If I wanna connect it from my office, what should I do? Do I need any port forwarding? Have you tried SSH VNC?

YH

作者: ckleea 時間: 2011-10-9 10:56

You need VNC, or NX for remote connection

作者: 角色 時間: 2011-10-9 11:00

本帖最後由角色於 2011-10-9 11:03 編輯

Thank you for your information.

For the time being, I just use

http://192.168.1.103:5081 to open it.

圖片附件: 10-9-2011 10-54-50 AM.png (2011-10-9 11:03, 89.52 KB) / 下載次數 588
http://telecom-cafe.com/forum/attachment.php?aid=945&k=f03d41ee79cf2ffe1366af49997638e2&t=1732317360&sid=HhgQHD

作者: 角色 時間: 2011-10-9 11:06

本帖最後由角色於 2011-10-9 11:07 編輯

Perhaps, we need more repositories shown below for CentOS such that they will include the rpm package for wine.

http://wiki.centos.org/Additiona ... direct=Repositories

http://wiki.centos.org/Additiona ... sitories/CentOSPlus

YH

作者: ckleea 時間: 2011-10-9 11:13

Two resposities are usually

Rpmforge and Epel

作者: 角色 時間: 2011-10-9 11:20

回復 22# ckleea

是的，我再找到一些资料。

http://24x7servermanagement.com/blog/?p=524

角色

作者: ckleea 時間: 2011-10-9 11:36

It should be more than enough

作者: 角色 時間: 2011-10-9 12:48

Yes, after having installed the wine package, I execute the command winecfg, I am able to have the following windows on my X window

角色

圖片附件: 10-9-2011 12-37-19 PM.png (2011-10-9 12:47, 6.85 KB) / 下載次數 693
http://telecom-cafe.com/forum/attachment.php?aid=946&k=236b7c59527281e8ae7f7fa92d274566&t=1732317360&sid=HhgQHD

作者: ckleea 時間: 2011-10-9 14:26

You are close to it.

作者: 角色 時間: 2011-10-9 15:03

在wine explorer，安装后OBiApp.exe

角色

圖片附件: 10-9-2011 2-51-30 PM.png (2011-10-9 15:02, 10.01 KB) / 下載次數 665
http://telecom-cafe.com/forum/attachment.php?aid=951&k=c15e463ceaf55aced7dbceec856e6da4&t=1732317360&sid=HhgQHD

作者: bubblestar 時間: 2011-10-12 23:39

回復 5# 角色

In your recommended website, they claimed that there is conflict between Skype and Wine. If you choose to install the Skype static version, it will also make Wine unavailable.

作者: 角色 時間: 2011-10-13 00:42

如果用Wine，就直接安装Window版本的Skype。

我估计应该可以可以。

角色

作者: ckleea 時間: 2011-10-13 06:32

You cannot make use of windows versions of skype to install multiple instances in the future. Wine is limited in functionality.

作者: ckleea 時間: 2011-10-13 06:38

Your way is towards installation of multiple instances of skype for > 1 channels to asterisk.

This is different from the OBiApps product that has only windows version only.

作者: bubblestar 時間: 2011-10-14 10:29

I am going to install siptosis in my system. Is it a good idea to install it with a general user instead of using root? How many instance can I make when running the free version?

Thanks

作者: ckleea 時間: 2011-10-14 12:46

I install as root user, same as asterisk

作者: bubblestar 時間: 2011-10-14 18:35

回復 33# ckleea

Following your advice, I reinstall siptosis using root as user. After installation, I tested my voice with Skype's built-in echo123 in GUI (gnome). However, only speakers work, microphone did not echo my voice recordings.

After rebooting the system, I did not see any Skype icon appear on Desktop right corner. Is it normal?

To verify Skype is running, I use the command 'top' in CLI and confirmed it is working. As I notice from http://wiki.centos.org/HowTos/Skype that there's a conflict between Skype and Wine (more precisely, between Skype and pulseaudio-libs.i686, on which Wine depends), would be plessed if you could tell how to resolve this?

For info., I have not created dialplan in Asterisk to test with Skype for the time being.

作者: ckleea 時間: 2011-10-14 23:33

回復 34# bubblestar

As spoken, as long as use linux static build skype, it should work. No worry on the audio in and out at the skype machine because you use it remotely. Concentrate to see if audio stream working when using asterisk to dial in and out

In my set up, I use siptosis to connect 9 skype users accounts and install wine + obiapps on the same machine. No conflict seen

For dial plan,

[CallingRule_Skype]; This serve 3 skype trunks _01 _02 _03 _05 _06 _07 _08 & _09 for outgoing to other skype accounts
exten => _83[1235679].,1,NoOp
exten => _83[1235679].,n,Dial(SIP/stsTrunk_0${EXTEN:2:1}/${EXTEN:3})
exten => _83[1235679].,n,Macro(stsdialresult)
exten => _83[1235679].,n,Playback(pls-try-call-later)
exten => _83[1235679].,n,Hangup()

複製代碼

[macro-stsdialresult]
; **** this is not complete - but a good start ****
; 603 Refused - hangup
; 404 Failed, Invalid user, no skype credit (Can't tell the difference) - hangup
; 408 UNPLACED whatever that means, try next channel
; 600 Busy - hangup
; 403 Anything else - hangup
;ISUP Cause value SIP response
; ---------------- ------------
; 1 unallocated number 404 Not Found
; 2 no route to network 404 Not found
; 3 no route to destination 404 Not found
; 16 normal call clearing --- (*)
; 17 user busy 486 Busy here
; 18 no user responding 408 Request Timeout
; 19 no answer from the user 480 Temporarily unavailable
; 20 subscriber absent 480 Temporarily unavailable
; 21 call rejected 403 Forbidden (+)
; 22 number changed (w/o diagnostic) 410 Gone
; 22 number changed (w/ diagnostic) 301 Moved Permanently
; 23 redirection to new destination 410 Gone
; 26 non-selected user clearing 404 Not Found (=)
; 27 destination out of order 502 Bad Gateway
; 28 address incomplete 484 Address incomplete
; 29 facility rejected 501 Not implemented
; 31 normal unspecified 480 Temporarily unavailable
exten => s,1,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS})
exten => s,2(debug1),Verbose(1,debug1 "${HANGUPCAUSE}:${DIALSTATUS}")
exten => s,3,Set(TIMEOUT(absolute)=120)
exten => s,4,GotoIf($[${HANGUPCAUSE} = 0]?s,6)
exten => s,5,Goto(cause-${HANGUPCAUSE},1)
exten => s,6,GotoIf($[${DIALSTATUS} = NOANSWER]?cause-19,1)
exten => s,7,GotoIf($[${DIALSTATUS} = BUSY]?cause-2,1)
exten => s,8,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?cause-0,1)
exten => s,9,GotoIf($[${DIALSTATUS} = ANSWER]?exit-1,1)
exten => s,10,Goto(cause-0,1)
exten => cause-0,1,NoOp(AST_CAUSE_NOTDEFINED)
exten => cause-0,n,Verbose(1,debug "cause-0")
exten => cause-0,n,Playback(error)
;exten => cause-0,n,Congestion
exten => cause-0,n,Goto(exit-1,1)
exten => cause-1,1,NoOp(AST_CAUSE_FAILURE)
exten => cause-1,n,Verbose(1,debug "cause-1 invalid destination")
exten => cause-1,n,Playback(invalid)
exten => cause-1,n,Hangup
exten => cause-2,1,NoOp(AST_CAUSE_BUSY)
exten => cause-2,n,Verbose(1,debug "cause-2 busy")
exten => cause-2,n,Busy
exten => cause-3,1,NoOp(AST_CAUSE_FAILURE)
exten => cause-3,n,Verbose(1,debug "cause-3")
;exten => cause-3,n,Playback(error)
exten => cause-3,n,Goto(exit-1,1)
exten => cause-4,1,NoOp(AST_CAUSE_CONGESTION)
exten => cause-4,n,Verbose(1,debug "cause-4")
exten => cause-4,n,Goto(exit-1,1)
exten => cause-5,1,NoOp(AST_CAUSE_UNALLOCATED)
exten => cause-5,n,Verbose(1,debug "cause-5 invalid destination")
exten => cause-5,n,Playback(invalid)
exten => cause-5,n,Hangup
exten => cause-18,1,NoOp(AST_CAUSE_CALL_UNPLACED)
exten => cause-18,n,Verbose(1,debug "cause-18 unplaced")
exten => cause-18,n,Goto(exit-1,1)
exten => cause-19,1,NoOp(AST_CAUSE_NO_ANSWER)
exten => cause-19,n,Verbose(1,debug "cause-19 noanswer")
exten => cause-19,n,Playback(noanswer)
exten => cause-19,n,Hangup
exten => cause-21,1,NoOp(AST_CAUSE_CALL_REJECTED)
exten => cause-21,n,Verbose(1,debug "cause-21 rejected")
exten => cause-21,n,Playback(rejected)
exten => cause-21,n,Hangup
exten => _cause-X,1,NoOp(UNKNOWN_CAUSECODE)
exten => _cause-X,n,Verbose(1,debug "cause-X")
exten => _cause-X,n,Playback(error)
;exten => _cause-X,n,Congestion
exten => _cause-X,n,Goto(exit-1,1)
exten => exit,1(exit),Noop

複製代碼

作者: bubblestar 時間: 2011-10-14 23:46

Thanks for the great dialplan example.

作者: ckleea 時間: 2011-10-15 00:03

回復 36# bubblestar

I can tell you that it is not difficult to make it running.

作者: bubblestar 時間: 2011-10-15 00:15

All dialplan codes are copied and put in place now. Will test it tomorrow morning. With the observation using 'top' in CLI environment, Skype has been running for more than 3 hours. Hope it can last without problem until I make the test.

I recall that you have a teachique to keep Skype running without being interrupted by the program per se. Lets discuss later for details. Thanks again for your great advice and assistance.

作者: ckleea 時間: 2011-10-15 07:07

I have the line in /etc/rc.d/rc.local

/opt/siptosis/stsTrunk_Control boot

作者: bubblestar 時間: 2011-10-15 11:44

I follow SipToSis's tutorial to install in /etc/rc.d/rc.local like this

su -l root -c "/root/siptosis/stsTrunk_linux boot"

I think our installation path is different but it should make no differences.

作者: bubblestar 時間: 2011-10-15 12:58

回復 35# ckleea

Could ckleea c-hing please share the Skype sip trunk settings in sip.conf for reference as well? I cannot figure out how to initiate the call without callee's info in the dialplan. Thanks

作者: 角色 時間: 2011-10-15 14:07

I guess bubblestar will eventually get the Skype Trunk working for both inbound and outbound Skype call.

YH

作者: ckleea 時間: 2011-10-15 20:31

This should be defined in sip.conf

[stsTrunk_01]
username=stsTrunk_01
type=friend
secret=yoursecret ; I don't kow how yours is generated
host=192.168.xxx.xxx
nat=no
dtmfmode=auto
canreinvite=no
port=5072
qualify=yes
defaultip=192.168.xxx.xxx
incominglimit=1
outgoinglimit=1
call-limit=1
busylevel=1

複製代碼

作者: 角色 時間: 2011-10-15 20:48

回復 43# ckleea

How about the configuration of the Sip2sis server?

How do we set the link between the number to Skype name?

YH

作者: bubblestar 時間: 2011-10-15 23:39

回復 43# ckleea

In fact, I did the similar settings in sip.conf this afternoon already. Thanks.

作者: ckleea 時間: 2011-10-16 07:54

回復 44# 角色

There are four config files need to set.

1. ststrunk.cfg
- need to check the port number and config files name for each trunk

2. siptoskypeauth.props
no change

3. skypetosipauth.props.
-define your incoming skype call to which asterisk extensions

4. skyeoutdialingrules.props.
-define your speed dial code

作者: ckleea 時間: 2011-10-16 07:56

Please show me the file names for the one channel version

作者: ckleea 時間: 2011-10-17 13:49

http://www.mhspot.com/sts/siptosis_skype_trunk_howto.html

Please follow also the the above link to install the trunk.

作者: bubblestar 時間: 2011-10-17 17:31

Both Skype and SipToSis are running. However, I still have registration problem between Asterisk and SipToSis. Typing "sip show peers" in CLI return "unreachable" remarks.

My sip.conf and siptosis.cfg are as below:

sip.conf in Asterisk

[skypetestuser]
username=skypetestuser ; use same as in brackets above
type=friend
context=default
secret=siptosisregpassword
host=192.168.888.888
port=5070
nat=yes
dtmfmode=auto
canreinvite=no
insecure = port,invite
qualify=yes
defaultip=192.168.888.888
incominglimit=1
outgoinglimit=1
call-limit=1
busylevel=1

複製代碼

siptosis.cfg in /opt/siptosis

#Sample Asterisk registration example - comment out NO registration info above first and uncomment the following.
host_port=5070
contact_url=sip:skypetestuser@192.168.888.888:5070
from_url="skypetestuser" <sip:skypetestuser@192.168.888.888:5060>
username=skypetestuser
realm=asterisk
passwd=siptosisregpassword
expires=3600
do_register=yes
minregrenewtime=120
regfailretrytime=15

複製代碼

作者: ckleea 時間: 2011-10-17 18:31

回復 49# bubblestar

What is the name of your second file? It is not the same as mine. My stsTrunk_01.cfg is much longer

作者: bubblestar 時間: 2011-10-17 23:48

回復 50# ckleea

I only have one siptosis.cfg file in the siptosis folder. There is no stsTrunk_01.cfg as mentioned in the directory. I think these other cfg files are generated only when Trunk Builder is used.

作者: ckleea 時間: 2011-10-18 06:36

回復 51# bubblestar
this is the nomenclature I used for multiple trunks. Please send me the files for a look

作者: bubblestar 時間: 2011-10-18 09:41

回復 52# ckleea

Send to you via email. Hope you can see the discrepancies. Thanks

作者: ckleea 時間: 2011-10-18 12:59

回復 53# bubblestar

See my version

Please note that I use _0X to denote individual trunk

#stsTrunkBuilder 20091019 generated trunk file 01 created: Tue Feb 15 17:04:58 HKT 2011 1527885965
logConfigFile=stsTrunk_01_log.properties
runConnectorReliabilityTest=no
skypeAPITrace=no
configWatchInterval=0
connectorWatchDogMinutes=0
connectionFee=0
MaxCallTimeLimitMinutes=0
WarnMinutesBeforeCutoff=1
OverLimitWarningFile=clips/overlimit.wav
OverUsageLimitSipResponse=480
dailyPstnLimitMinutes=350
dailyPstnUniqueNumberLimit=48
refuseNewPstnCallsWhenRemainingMinutesUnder=5
MaxPstnCallTimeLimitMinutes=0
loadSkypeClientCallHistory=yes
tollFreeNumberPrefixes=1800,1888,1866,1877
emailWhenBalanceDropsTo=-1
emailHost=
emailPort=25
emailusername=stsTrunk_01
emailPassword=
emailRecipients=
emailFrom=
emailTest=no
setSkypeOnlineStatusInterval=0
skypeOnlineStatus=ONLINE
callBackForceSipPrefix=*
callLogPath=log/
siptoskypeauthfile=SipToSkypeAuth_01.props
skypetosipauthfile=SkypeToSipAuth_01.props
SkypeOutDialingRulesFile=SkypeOutDialingRules_01.props
SipOutDialingRulesFile=SipOutDialingRules_01.props
ua_jar=ua.jar
audioPriorityIncrease=0
jitterLevel=2
skype_connect=yes
skype_audioportbase=64436
enableSkypeDtmfDetector=yes
SkypeDtmfDetectorHitThreshold=40
SkypeDtmfDetectorSilenceThreshold=6
sendSipDtmfToSkype=yes
sendSkypeDtmfToSip=yes
inbandFullTimeDtmfDetection=yes
JoinManualSkypeOutboundCallToSip=no
SkypeInboundAllChannelsBusyAction=transferto:[color=Red]your_skype_username[/color]
SkypeTransferTimeoutMs=8000
SkypeInboundSipDestUnavailableAction=refuse
SipInboundAllChannelsBusyAction=busy
skypeclientsupportsmulticalls=no
concurrentcalllimit=1
autoShutdownMinutes=0
pintimeout=8
pinretrylimit=3
destinationtimeout=12
destinationretrylimit=3
pinFile=clips/enterPin.wav
destinationFile=clips/enterDest.wav
dialingFile=clips/dialing.wav
invalidPinFile=clips/invalidPin.wav
invalidDestFile=clips/invalidDest.wav
skypePinFile=clips/enterPin.wav
skypeDestinationFile=clips/enterDest.wav
skypeDialingFile=clips/dialing.wav
skypeInvalidPinFile=clips/invalidPin.wav
skypeInvalidDestFile=clips/invalidDest.wav
handleEarlyMedia=yes
handleSipEarlyMedia=no
sendRingToSkypeCaller=no
skypeRingFile=clips/skypeRing.wav
skypeRingInterval=8
sendSkypeEarlyMediaOverSipSessionProgress=yes
replaceFromWithSkypeId=no
sendSkypeIM=no
skypeimmessage=You are about to receive a Skype Voice call from [callerid] [callernumber].
sendSkypeImDelay=2
transport_protocols=udp
stunTestInterval=30
enableNatTranslate=yes
enableNatTranslateVia=no
host_port=5072
username=stsTrunk_01
passwd=your_generated_passkey
do_register=no
keepalive_time=0
audio=yes
audio_port=63204
noRtpReceivedAutoHangupSeconds=30
audio_codec=PCMU,PCMA,ILBC
audio_frame_size=240,240,240
audio_avp=-1,-1,98
skype_audiooutgain=1,1,1
skype_audioingain=1.5,1.5,1.5
FilterParams=NONE
enableSendRTPtoReceivedAddress=yes
lockRtpSendAddressAfterPackets=1
dtmf2833payloadtype=101
enableSIPInbandDtmfDetector=no
SipDtmfDetectorHitThreshold=30
SipDtmfDetectorSilenceThreshold=6
useViaRport=yes
useViaReceived=yes
sendResponseUsingOutboundProxy=no
baseFailureResponse=403
skypeRefusedResponse=603
skypeFailedResponse=404
skypeUnPlacedResponse=408
skypeBusyResponse=600
TcpRxBufferSize=8192
TcpTxBufferSize=8192
RtpRxBufferSize=8192
RtpTxBufferSize=8192
is_registrar=yes
register_new_users=yes
allowMultiContactsPerUser=no

複製代碼

作者: bubblestar 時間: 2011-10-19 00:31

Wow! Such a long script. I must work hard to digest the content. Thanks for your help in advance.

作者: ckleea 時間: 2011-10-19 08:06

回復 55# bubblestar
大部分都是跟 default，慢慢試。

作者: bubblestar 時間: 2011-10-20 23:53

本帖最後由 bubblestar 於 2011-10-20 23:56 編輯

SipToSis 是否已完全變成免費版呢?? 因為收費版本的Download 不見了(被剷走)，如果係，真是好消息。

http://www.mhspot.com/sts/siptosis_download.php

作者: 角色 時間: 2011-10-21 00:13

是的，现在开始安装Skype for Linux （static）。

角色

作者: ckleea 時間: 2011-10-21 06:26

回復 57# bubblestar

Somethings must be changed

作者: ckleea 時間: 2011-10-21 06:27

Please also note

Call-Back Setup Instructions

Note: PSTN rates will be per PSTN outbound call according to the selected provider billing terms.

Single Stage Callback - no IVR or additional dialing - Scroll down for two stage metbod.

Trigger callback using a Skype DID (AKA SkypeIn,Skype Online number) or from a Specific Skype User
Edit SkypeToSipAuth.props
Add a line like this (at least two targets):
AuthorizedSkypeIdOrNumber,CallBack:Skype=someid1OrPstnNumber,someskypeid2OrPstnNumber
or:
AuthorizedSkypeIdOrNumber,CallBack:Skype=someid1OrPstnNumber|SIP=someSipAddress@someprovider:5060
Note: only a single SIP target can be specified.

Using a SIP DID to trigger callback
Edit SipToSkypeAuth.props
Add a line like this (at least two targets):
AuthorizedSIPNumber,*,*,CallBack:Skype=someSkypeId1OrPstnNumber,someSkypeId2OrPstnNumber
or:
AuthorizedSIPNumber,*,*,CallBack:Skype=someSkypeId1,someSkypeId2|SIP=someSipAddress@someprovider:5060
Note: only a single SIP target can be specified.

Another way of using a SIP call to trigger callback
Edit SkypeOutDialingRules.props
Add a line like this (at least two targets):
^58$:CallBack:Skype=someskypeuser1OrPstnNumber,someskypeuser2OrPstnNumber
or:
^58$:CallBack:Skype=someskypeuser1OrPstnNumber,someskypeuser2OrPstnNumber|SIP=someSIPUser@SomeSIPAddress:5060
In this example, if you dial 58@yourSTSGateway:stsPort - it will trigger a callback.
Note: only a single SIP target can be specified.

Two Stage Callback - uses IVR for dialing
Note: DTMF decoding must be on. In the case of a Skype PSTN target, DTMF decoding may not be reliable.

Trigger callback using a Skype DID (AKA SkypeIn,Skype Online number) or from a Specific Skype User
Edit SkypeToSipAuth.props
Add a line like this (specify only one target):
AuthorizedSkypeIdOrNumber,CallBack:Skype=someSkypeIdOrPSTNNumber
or:
AuthorizedSkypeIdOrNumber,CallBack:SIP=someSipAddress@someprovider:5060
Once the called back target answers, the IVR will prompt for the destination.
Destination will be dialed as defined in SkypeOutDialingRules.props.
Default is to call out via Skype, to dial out using SIP instead, dial * before the destination.
Parameter callBackForceSipPrefix controls the SIP dialing prefix.
In the case of SIP dialing, destination will be dialed as defined in SipOutDialingRules.

Using a SIP DID to trigger callback
Edit SipToSkypeAuth.props
Add a line like this (specify only one target):
AuthorizedSIPNumber,*,*,CallBack:Skype=someSkypeIdOrPSTNNumber
or:
AuthorizedSIPNumber,*,*,CallBack:SIP=someSipAddress@someprovider:5060
Once the called back target answers, the IVR will prompt for the destination.
Destination will be dialed as defined in SkypeOutDialingRules.props.

Another way of using a SIP call to trigger callback
Edit SkypeOutDialingRules.props
Add a line like this (specify only one target):
^58$:CallBack:Skype=someSkypeIdOrPSTNNumber
or:
^58$:CallBack:SIP=someSIPUser@SomeSIPAddress:5060
In this example, if you dial 58@yourSTSGateway:stsPort - it will trigger a callback.
Once the called back target answers, the IVR will prompt for the destination.
Destination will be dialed as defined in SkypeOutDialingRules.props.

作者: ckleea 時間: 2011-10-22 22:41

To enable all on status, go to the config.xml and change

<IdleTimeForAway>0</IdleTimeForAway>
<IdleTimeForNA>0</IdleTimeForNA>

Then your skype status won't be away

作者: bubblestar 時間: 2011-10-22 23:01

Thanks for the useful information.

作者: ckleea 時間: 2011-10-27 08:00

For your information, i believe the paid version of stsTrunkbuilder has been discontinued and withdrawn from usage.

Though i download the free version from the site, the content is completely different from paid version that i have.

作者: bubblestar 時間: 2011-10-27 12:54

Have you tried installing the free version of stsTrunkbuilder and compare it with your paid version about their functionality?

作者: 角色 時間: 2011-10-27 20:48

Looking forward to seeing the difference.

YH

作者: ckleea 時間: 2011-10-27 22:03

回復 65# 角色

I look at the props and cfg files. There have been more settings in the new version.
I am unable to replace my old files with the new one as it would overwrite mine. What I did before, use trunkbuilder to build the trunk settings, then copy to sip.conf. Config each skype instance to accept skype-java and auto login etc.

Then autoboot with /etc/rc.d/rc.local

作者: 角色 時間: 2011-10-30 08:01

The installation is almost completed. The next task may be the installation of multiple Skype clients.

作者: fatfish 時間: 2011-10-30 09:36

过两天我也要用这个帖子了
哈哈哈哈

作者: 角色 時間: 2011-10-30 21:53

You are welcome to have a look at this thread when you install the Skype for Asterisk.

YH

作者: ckleea 時間: 2011-11-3 03:47

Siptosis has a lot of settings needed to look at. It seems very unfortunate that the developer has taken out the paid version from the site and will provide limited support only for paid users.
I am looking forward for an upgrade.

作者: ckleea 時間: 2011-11-3 03:49

I have just tried this combination

skype in US -> siptosis on asterisk in HK -> sip phone in UK

Almost like the usual copper wire analog phone calls.

作者: ckleea 時間: 2011-11-5 19:50

其實 siptosis.cfg內有很多 setting 可以調控，always online is one, watchdog on configuration changes

但運作上，有時會出現 trunk not reachable.，原因不明?

作者: ckleea 時間: 2011-11-9 07:43

回復 61# ckleea

If you are not using stsTrunkBuilder, you may not be able to find the setting.

Another workaround to keep your skype status online is to change your stsTrunk.cfg, edit as

setSkypeOnlineStatusInterval=1
skypeOnlineStatus=ONLINE

If you want the have the config change while you are already online, edit the following
configWatchInterval=1
connectorWatchDogMinutes=1

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