【角色茶桌】——Asterisk 1.8 + Siptosis彻底安装成功！

角色

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

1. tar jxvf skype_static-2.1.0.81.tar.bz2

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to /usr/src/skype directory

3. Create symbolic links

1. ln -s /usr/src/skype /usr/share/skype
   
2. ln -s /usr/src/skeype/skype /usr/bin/skype

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4. Install X Window and other packages

1. yum install libXv
   
2. yum install libXScrnSaver
   
3. yum groupinstall "X Window System"
   
4. yum install alsa-lib

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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:

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

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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

1. su -l root
   
2. [root-password]
   
3. 
   
4. cd /usr/src
   
5. 
   
6. // To start X Window
   
7. startx
   
8. 
   
9. // To start Skype
   
10. skype

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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

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

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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)

1. [skypests]
   
2. username=skypests
   
3. type=friend
   
4. secret=skype
   
5. host=192.168.1.103
   
6. nat=no
   
7. dtmfmode=auto
   
8. ;canreinvite=yes (use only if you understand what it does - does not work well with ilbc and speex codecs)
   
9. canreinvite=no
   
10. ;port should not be needed if you register with the PBX - some have said it's needed??
    
11. ;port=siptosishostport
    
12. port=5070
    
13. qualify=yes
    
14. defaultip=192.168.1.103
    
15. incominglimit=1
    
16. outgoinglimit=1
    
17. call-limit=1
    
18. busylevel=1
    
19. context=from-skype

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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.

1. #you can simulate speed dials this way also (dialing your prefix and 55 would call the skype echo test)
   
2. ^55$:echo123
   
3. ^56$:skype_name_1
   
4. ^57$:skype_name_2
   
5. #send callme im to echo123 to get a call back from the test service
   
6. ^559$:im:echo123:callme

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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.

1. #*,sip:3000@192.168.1.103:5228
   
2. *,sip:1911@192.168.1.5:5060
   
3. 
   
4. #Default: all incoming skype callers get the invalid destination message
   
5. *,play:clips/invalidDest.wav

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角色

ckleea

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

ckleea

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?

ckleea

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.

ckleea

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.

角色

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

YH

fatfish

过两天我也要用这个帖子了
哈哈哈哈

角色

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

ckleea

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

角色

Looking forward to seeing the difference.

YH

bubblestar

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

ckleea

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.

bubblestar

Thanks for the useful information.

ckleea

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

ckleea

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.