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標題: Skype to Asterisk-GUI via Windows siptosis [打印本頁]
作者: 角色    時間: 2013-2-12 19:11     標題: Skype to Asterisk-GUI via Windows siptosis

本帖最後由 角色 於 2013-2-12 19:31 編輯

Skype to Asterisk via Linux siptosis在其他文章已经说了很多，但是siptosis在Windows下，怎样连接Asterisk-GUI呢就很少文章提及过。希望通过下面的文章可以教大家怎样利用Windows siptosis，把Skype连接到Asterisk-GUI server里。

目的：用Skype Client拨打另外一个Skype Client，然后连到另外一个Asterisk（-GUI）Server，再通过IVR接到其他extensions，落地trunk。

为什么要用Skype？你可以可以用SIP Client，但是很多地方对SIP Client有限制，很多时候都打不通，再加上codec问题，导致不太流畅。特别在网络环境不太好，VoIP blocking的地区为甚的地方。而Skype有本身的自己的SILK codec，在这种不太好的情况下也能发挥得非常好！

坊间有不少的Skype-to-Asterisk的gateway，但是以siptosis最为简单而易用，最重要是免费。
作者: 角色    時間: 2013-2-12 19:11

本帖最後由 角色 於 2013-2-12 19:50 編輯

我们所需要的软件：

软件；
1. Siptosis ：http://www.mhspot.com/sts/siptosis_download.php
2. Skype Client (Windows) ：www.skype.com
3. Java JRE : JRE6
4. Asterisk-GUI Server (see other thread)
作者: 角色    時間: 2013-2-12 19:11

本帖最後由 角色 於 2013-2-12 21:00 編輯

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

1. 下载SipToSis software

1003.gif

2. 然后把它放在适合的位置，再解压，里面的contents：

1004.gif

3. Read the readme：
  1. Copyright (C) 2009 Greg Dorfuss - mhspot.com

  3. SipToSis is free software; you can redistribute it and/or modify
  4. it under the terms of the GNU General Public License as published by
  5. the Free Software Foundation; either version 3 of the License, or
  6. (at your option) any later version.

  8. SipToSis is distributed in the hope that it will be useful,
  9. but WITHOUT ANY WARRANTY; without even the implied warranty of
  10. MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  11. GNU General Public License for more details.

  13. You should have received a copy of the GNU General Public License
  14. along with this source code; if not, write to the Free Software
  15. Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

  17. Based on mjsip 1.6 software.
  18. Uses skype4java and log4j libraries.
  19. mjsip, skype4java and log4j have their own license agreements.

  21. This product uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype

  23. Installation:

  25. Install SUN's java 1.5 or higher for your platform - make sure it is available in the system path.
  26. OSX MUST use 32 bit java since there is no 64 bit OSX native connector. Windows and Linux can use either 32 or 64 bit.
  27. You can check the version from a command prompt like this: java -version


  30. For new install:
  31.     Windows/Linux:
  32.            Unzip the siptosis archive into a folder called siptosis or other folder name.

  34. Startup Skype (must be the same computer).

  36. Windows:
  37.         double click SipToSis_win.bat
  38.         If you get warning about JAVAEXEPATH on Windows 64 bit with 64 bit Java:
  39.           edit SipToSis_win.bat and uncomment the line "set JAVAEXEPATH" - make sure it points to your JRE or JDK bin folder.
  40.        
  41. The Skype program may ask if access should be allowed. If you don't authorize it, it will not work.
  42. You usually have to go into the Skype client and manually authorize it in the options screen
  43. (Even though you clicked allow access - some sort of skype bug - depends on skype version).
  44. Your Skype User Id should appear on the console. If not stop and fix problem.

  46. If the SipToSis console shows a contact_url of 127.0.0.1 instead of the machine's actual IP address,
  47. edit the siptosis.cfg file, comment out the "#Sample AUTO config with NO registration" section
  48. and setup the "#Sample config with NO registration" section by setting 127.0.0.1 to the SipToSis machine's IP address
  49. and also set the via_addr. If you know what you are doing, you can enable one of the other registration sections instead.


  52. Now setup your VOIP/SIP adapter or PBX to call the SipToSis Computer.

  54. SPA-3102 Dial Plan Example and other Linksys sipura devices:
  55.         (<#1:>xx.<:@siptosisIpAddress:siptosisHostPort>|originaldialplan)
  56.         At this point, dialing #155 should get you the Skype Echo Test service.
  57.         Only the Skype Echo Test service and PSTN is callable at this point.
  58.         You will customize this later on.
  59.        
  60. Asterisk Peer Example:
  61. In sip.conf       
  62. [siptosisuser]
  63. username=siptosisuser
  64. type=friend
  65. context=default
  66. secret=siptosisregpassword
  67. host=dynamic
  68. nat=no
  69. dtmfmode=auto
  70. ;canreinvite=yes (use only if you understand what it does - does not work well with ilbc and speex codecs)
  71. canreinvite=no
  72. ;port should not be needed if you register with the PBX - some have said it's needed??
  73. ;port=siptosishostport
  74. qualify=yes
  75. defaultip=siptosisip
  76. incominglimit=1
  77. outgoinglimit=1
  78. call-limit=1
  79. busylevel=1

  81. Asterisk Single channel config that can be put in extensions.conf
  82. exten => _7X.,1,Dial(SIP/siptosisuser/${EXTEN:1})
  83. ;you would then dial 7 and the number you want to call

  85. ** Customize and authorize for regular dialing:
  86.    Assign speed dial numbers in the Skype client to each Skype contact you want to call.
  87.    or create mappings in SipToSkypeAuths.props.
  88.    You don't need to do this if using only skypeout.
  89.    If using a PBX you could map extensions to Skype users there instead.

  91.    Edit SipToSkypeAuth.props to forward and authorize SIP calls to desired Skype destinations. (Most users will only use: *,*,localnet,calleeid)
  92.    Edit SkypeOutDialingRules.props for any Skype dialing rules/transforms wanted.
  93.    Edit 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 message).

  95. At this point, you should be able to make and receive skype calls. The remainder of this document contains
  96. some optional configuration and troubleshooting instructions. More documentation is available on http://mhspot.com


  99. CallBack config:
  100.    *** CHECK LEGALITY OF USING CALLBACK FOR YOUR COUNTRY *** USE AT YOUR OWN RISK ***
  101.    Compatability:
  102.      Windows client 4.1.0.141 works fine.
  103.    Callbacks can be initiated from SIP and Skype DIDs assuming Caller Id works with the DID or can be initiated from Skype User.
  104.    See sample files SipToSkypeAuth.props,SkypeOutDialingRules.props,SkypeToSipAuth.props for examples.
  105.    Two stage dialing requires the SkypeInBand decoder turned on:
  106.       enableSkypeDtmfDetector=yes
  107.       inbandFullTimeDtmfDetection=yes

  109. Conference Calls:
  110.    Compatability:
  111.      Windows client 4.1.0.141 works fine.
  112.    See sample files SipToSkypeAuth.props and SkypeOutDialingRules.props for examples.

  114. Codec Configuration:
  115.   GSM Codec:
  116.     To use Tritonus download tritonus_gsm-0.3.6.jar and tritonus_share-0.3.6.jar from tritonus.org and put in same folder.
  117.     To use Sun's JMF , download Sun's Java Media Framework and put jmf.jar in same folder.
  118.   PCMU(u-law),PCMA(a-law),iLBC Codecs
  119.     Already built in

  121.   See the siptosis.cfg file and look for audio_codec= to configure codecs and sound volume.

  123. You can specify a different config file like this:
  124.    SipToSis_win.bat myotherconfig.cfg       

  126. For a multi channel trunk, you might want to try stsTrunkBuilder.
  127.    
  128. To see full debug information set the following in siptosis.cfg
  129. logConfigFile=log_debug.properties
  130. and for Skype API tracing:
  131. skypeAPITrace=yes

  133. On Windows:
  134.      If you get the unsatisfied link error "Can't load IA 32-bit .dll on a AMD 64-bit platform"
  135.      delete skype.dll and let the installer copy the correct one in.
複製代碼


圖片附件: 1003.gif (2013-2-12 19:59, 31.98 KB) / 下載次數 776
http://telecom-cafe.com/forum/attachment.php?aid=2212&k=68cd9093c62c61a62c9c680ba68f74bd&t=1732359710&sid=p6orP5



圖片附件: 1004.gif (2013-2-12 20:51, 35.82 KB) / 下載次數 774
http://telecom-cafe.com/forum/attachment.php?aid=2213&k=849710e086a36754f58be62ab84f9132&t=1732359710&sid=p6orP5

作者: 角色    時間: 2013-2-12 19:11

本帖最後由 角色 於 2013-2-12 20:59 編輯

readme content是：
  1. Copyright (C) 2009 Greg Dorfuss - mhspot.com

  3. SipToSis is free software; you can redistribute it and/or modify
  4. it under the terms of the GNU General Public License as published by
  5. the Free Software Foundation; either version 3 of the License, or
  6. (at your option) any later version.

  8. SipToSis is distributed in the hope that it will be useful,
  9. but WITHOUT ANY WARRANTY; without even the implied warranty of
  10. MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  11. GNU General Public License for more details.

  13. You should have received a copy of the GNU General Public License
  14. along with this source code; if not, write to the Free Software
  15. Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

  17. Based on mjsip 1.6 software.
  18. Uses skype4java and log4j libraries.
  19. mjsip, skype4java and log4j have their own license agreements.

  21. This product uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype

  23. Installation:

  25. Install SUN's java 1.5 or higher for your platform - make sure it is available in the system path.
  26. OSX MUST use 32 bit java since there is no 64 bit OSX native connector. Windows and Linux can use either 32 or 64 bit.
  27. You can check the version from a command prompt like this: java -version


  30. For new install:
  31.     Windows/Linux:
  32.            Unzip the siptosis archive into a folder called siptosis or other folder name.

  34. Startup Skype (must be the same computer).

  36. Windows:
  37.         double click SipToSis_win.bat
  38.         If you get warning about JAVAEXEPATH on Windows 64 bit with 64 bit Java:
  39.           edit SipToSis_win.bat and uncomment the line "set JAVAEXEPATH" - make sure it points to your JRE or JDK bin folder.
  40.        
  41. The Skype program may ask if access should be allowed. If you don't authorize it, it will not work.
  42. You usually have to go into the Skype client and manually authorize it in the options screen
  43. (Even though you clicked allow access - some sort of skype bug - depends on skype version).
  44. Your Skype User Id should appear on the console. If not stop and fix problem.

  46. If the SipToSis console shows a contact_url of 127.0.0.1 instead of the machine's actual IP address,
  47. edit the siptosis.cfg file, comment out the "#Sample AUTO config with NO registration" section
  48. and setup the "#Sample config with NO registration" section by setting 127.0.0.1 to the SipToSis machine's IP address
  49. and also set the via_addr. If you know what you are doing, you can enable one of the other registration sections instead.


  52. Now setup your VOIP/SIP adapter or PBX to call the SipToSis Computer.

  54. SPA-3102 Dial Plan Example and other Linksys sipura devices:
  55.         (<#1:>xx.<:@siptosisIpAddress:siptosisHostPort>|originaldialplan)
  56.         At this point, dialing #155 should get you the Skype Echo Test service.
  57.         Only the Skype Echo Test service and PSTN is callable at this point.
  58.         You will customize this later on.
  59.        
  60. Asterisk Peer Example:
  61. In sip.conf       
  62. [siptosisuser]
  63. username=siptosisuser
  64. type=friend
  65. context=default
  66. secret=siptosisregpassword
  67. host=dynamic
  68. nat=no
  69. dtmfmode=auto
  70. ;canreinvite=yes (use only if you understand what it does - does not work well with ilbc and speex codecs)
  71. canreinvite=no
  72. ;port should not be needed if you register with the PBX - some have said it's needed??
  73. ;port=siptosishostport
  74. qualify=yes
  75. defaultip=siptosisip
  76. incominglimit=1
  77. outgoinglimit=1
  78. call-limit=1
  79. busylevel=1

  81. Asterisk Single channel config that can be put in extensions.conf
  82. exten => _7X.,1,Dial(SIP/siptosisuser/${EXTEN:1})
  83. ;you would then dial 7 and the number you want to call

  85. ** Customize and authorize for regular dialing:
  86.    Assign speed dial numbers in the Skype client to each Skype contact you want to call.
  87.    or create mappings in SipToSkypeAuths.props.
  88.    You don't need to do this if using only skypeout.
  89.    If using a PBX you could map extensions to Skype users there instead.

  91.    Edit SipToSkypeAuth.props to forward and authorize SIP calls to desired Skype destinations. (Most users will only use: *,*,localnet,calleeid)
  92.    Edit SkypeOutDialingRules.props for any Skype dialing rules/transforms wanted.
  93.    Edit 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 message).

  95. At this point, you should be able to make and receive skype calls. The remainder of this document contains
  96. some optional configuration and troubleshooting instructions. More documentation is available on http://mhspot.com


  99. CallBack config:
  100.    *** CHECK LEGALITY OF USING CALLBACK FOR YOUR COUNTRY *** USE AT YOUR OWN RISK ***
  101.    Compatability:
  102.      Windows client 4.1.0.141 works fine.
  103.    Callbacks can be initiated from SIP and Skype DIDs assuming Caller Id works with the DID or can be initiated from Skype User.
  104.    See sample files SipToSkypeAuth.props,SkypeOutDialingRules.props,SkypeToSipAuth.props for examples.
  105.    Two stage dialing requires the SkypeInBand decoder turned on:
  106.       enableSkypeDtmfDetector=yes
  107.       inbandFullTimeDtmfDetection=yes

  109. Conference Calls:
  110.    Compatability:
  111.      Windows client 4.1.0.141 works fine.
  112.    See sample files SipToSkypeAuth.props and SkypeOutDialingRules.props for examples.

  114. Codec Configuration:
  115.   GSM Codec:
  116.     To use Tritonus download tritonus_gsm-0.3.6.jar and tritonus_share-0.3.6.jar from tritonus.org and put in same folder.
  117.     To use Sun's JMF , download Sun's Java Media Framework and put jmf.jar in same folder.
  118.   PCMU(u-law),PCMA(a-law),iLBC Codecs
  119.     Already built in

  121.   See the siptosis.cfg file and look for audio_codec= to configure codecs and sound volume.

  123. You can specify a different config file like this:
  124.    SipToSis_win.bat myotherconfig.cfg       

  126. For a multi channel trunk, you might want to try stsTrunkBuilder.
  127.    
  128. To see full debug information set the following in siptosis.cfg
  129. logConfigFile=log_debug.properties
  130. and for Skype API tracing:
  131. skypeAPITrace=yes

  133. On Windows:
  134.      If you get the unsatisfied link error "Can't load IA 32-bit .dll on a AMD 64-bit platform"
  135.      delete skype.dll and let the installer copy the correct one in.
複製代碼
作者: 角色    時間: 2013-2-12 19:11

备用帖子。。。
作者: 角色    時間: 2013-2-12 19:32

备用帖子。。。
作者: 角色    時間: 2013-2-12 19:34

备用帖子。。。
作者: 角色    時間: 2013-2-12 19:35

备用帖子。。。
作者: 角色    時間: 2013-2-12 19:35

备用帖子。。。
作者: 角色    時間: 2013-2-12 19:35

备用帖子。。。
作者: 雯雯    時間: 2013-2-12 19:44

我想我知道其中1個原因了, 我沒有在VM裏set sound card.
作者: 角色    時間: 2013-2-12 19:51

回復 11# 雯雯

其实你要再VM上，看是你用Skype Client是否能城市使用呢？这个非常关键！
作者: 浮雲1965    時間: 2013-3-20 16:14

請問角色兄：
是不是“Skype to Asterisk-GUI via Windows siptosis” 是需要一臺電腦跑windows, 然后在上面跑siptosis ?
因為我們的Elastix是在linux上的，那是不是如果用 “Skype to Asterisk-GUI via Linux siptosis”, 就可以在架設Elastix的同一臺機器上跑siptosis? 從而完成Skype的接入？

謝謝！
作者: ckleea    時間: 2013-3-20 17:56

用 linux 會方便 D
作者: 浮雲1965    時間: 2013-3-20 19:38

用 linux 會方便 D
ckleea 發表於 2013-3-20 17:56



我在壇內找到一些教程，不知道是否update? 這個siptosis還是免費的嗎？

謝謝！
作者: 角色    時間: 2013-3-20 22:29

回復 15# 浮雲1965

Siptosis可以在Linux运行都没有问题，但是要安装JRE。



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