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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.

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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

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I install as root user, same as asterisk

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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.

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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,
  1. [CallingRule_Skype]; This serve 3 skype trunks _01 _02 _03 _05 _06 _07 _08 & _09 for outgoing to other skype accounts
  2. exten => _83[1235679].,1,NoOp
  3. exten => _83[1235679].,n,Dial(SIP/stsTrunk_0${EXTEN:2:1}/${EXTEN:3})
  4. exten => _83[1235679].,n,Macro(stsdialresult)
  5. exten => _83[1235679].,n,Playback(pls-try-call-later)
  6. exten => _83[1235679].,n,Hangup()
複製代碼
  1. [macro-stsdialresult]
  2. ; **** this is not complete - but a good start ****
  3. ;       603 Refused - hangup
  4. ;       404 Failed, Invalid user, no skype credit (Can't tell the difference) - hangup
  5. ;       408 UNPLACED whatever that means, try next channel
  6. ;       600 Busy - hangup
  7. ;       403 Anything else - hangup

  8. ;ISUP Cause value                        SIP response
  9. ;  ----------------                        ------------
  10. ;  1  unallocated number                   404 Not Found
  11. ;  2  no route to network                  404 Not found
  12. ;  3  no route to destination              404 Not found
  13. ;  16 normal call clearing                 --- (*)
  14. ;  17 user busy                            486 Busy here
  15. ;  18 no user responding                   408 Request Timeout
  16. ;  19 no answer from the user              480 Temporarily unavailable
  17. ;  20 subscriber absent                    480 Temporarily unavailable
  18. ;  21 call rejected                        403 Forbidden (+)
  19. ;  22 number changed (w/o diagnostic)      410 Gone
  20. ;  22 number changed (w/ diagnostic)       301 Moved Permanently
  21. ;  23 redirection to new destination       410 Gone
  22. ;  26 non-selected user clearing           404 Not Found (=)
  23. ;  27 destination out of order             502 Bad Gateway
  24. ;  28 address incomplete                   484 Address incomplete
  25. ;  29 facility rejected                    501 Not implemented
  26. ;  31 normal unspecified                   480 Temporarily unavailable
  27. exten => s,1,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS})
  28. exten => s,2(debug1),Verbose(1,debug1 "${HANGUPCAUSE}:${DIALSTATUS}")
  29. exten => s,3,Set(TIMEOUT(absolute)=120)
  30. exten => s,4,GotoIf($[${HANGUPCAUSE} = 0]?s,6)
  31. exten => s,5,Goto(cause-${HANGUPCAUSE},1)
  32. exten => s,6,GotoIf($[${DIALSTATUS} = NOANSWER]?cause-19,1)
  33. exten => s,7,GotoIf($[${DIALSTATUS} = BUSY]?cause-2,1)
  34. exten => s,8,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?cause-0,1)
  35. exten => s,9,GotoIf($[${DIALSTATUS} = ANSWER]?exit-1,1)
  36. exten => s,10,Goto(cause-0,1)

  37. exten => cause-0,1,NoOp(AST_CAUSE_NOTDEFINED)
  38. exten => cause-0,n,Verbose(1,debug "cause-0")
  39. exten => cause-0,n,Playback(error)
  40. ;exten => cause-0,n,Congestion
  41. exten => cause-0,n,Goto(exit-1,1)

  42. exten => cause-1,1,NoOp(AST_CAUSE_FAILURE)
  43. exten => cause-1,n,Verbose(1,debug "cause-1 invalid destination")
  44. exten => cause-1,n,Playback(invalid)
  45. exten => cause-1,n,Hangup

  46. exten => cause-2,1,NoOp(AST_CAUSE_BUSY)
  47. exten => cause-2,n,Verbose(1,debug "cause-2 busy")
  48. exten => cause-2,n,Busy

  49. exten => cause-3,1,NoOp(AST_CAUSE_FAILURE)
  50. exten => cause-3,n,Verbose(1,debug "cause-3")
  51. ;exten => cause-3,n,Playback(error)
  52. exten => cause-3,n,Goto(exit-1,1)

  53. exten => cause-4,1,NoOp(AST_CAUSE_CONGESTION)
  54. exten => cause-4,n,Verbose(1,debug "cause-4")
  55. exten => cause-4,n,Goto(exit-1,1)

  56. exten => cause-5,1,NoOp(AST_CAUSE_UNALLOCATED)
  57. exten => cause-5,n,Verbose(1,debug "cause-5 invalid destination")
  58. exten => cause-5,n,Playback(invalid)
  59. exten => cause-5,n,Hangup

  60. exten => cause-18,1,NoOp(AST_CAUSE_CALL_UNPLACED)
  61. exten => cause-18,n,Verbose(1,debug "cause-18 unplaced")
  62. exten => cause-18,n,Goto(exit-1,1)

  63. exten => cause-19,1,NoOp(AST_CAUSE_NO_ANSWER)
  64. exten => cause-19,n,Verbose(1,debug "cause-19 noanswer")
  65. exten => cause-19,n,Playback(noanswer)
  66. exten => cause-19,n,Hangup

  67. exten => cause-21,1,NoOp(AST_CAUSE_CALL_REJECTED)
  68. exten => cause-21,n,Verbose(1,debug "cause-21 rejected")
  69. exten => cause-21,n,Playback(rejected)
  70. exten => cause-21,n,Hangup

  71. exten => _cause-X,1,NoOp(UNKNOWN_CAUSECODE)
  72. exten => _cause-X,n,Verbose(1,debug "cause-X")
  73. exten => _cause-X,n,Playback(error)
  74. ;exten => _cause-X,n,Congestion
  75. exten => _cause-X,n,Goto(exit-1,1)

  76. exten => exit,1(exit),Noop
複製代碼

TOP

Thanks for the great dialplan example.

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回復 36# bubblestar


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

TOP

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.

TOP

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

/opt/siptosis/stsTrunk_Control boot

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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.

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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

TOP

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

YH

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This should be defined in sip.conf
  1. [stsTrunk_01]
  2. username=stsTrunk_01
  3. type=friend
  4. secret=yoursecret ; I don't kow how yours is generated
  5. host=192.168.xxx.xxx
  6. nat=no
  7. dtmfmode=auto
  8. canreinvite=no
  9. port=5072
  10. qualify=yes
  11. defaultip=192.168.xxx.xxx
  12. incominglimit=1
  13. outgoinglimit=1
  14. call-limit=1
  15. busylevel=1
複製代碼

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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

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回復 43# ckleea


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

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