ckleea

Please describe a bit your set up.

If you use asterisk, can you see siptosis client online by typing sip show peers in cli?

The siptosis setup required detailed reading of instruction.

ckleea

回復 7# kurtor

So you can receive skype in call at sip client? If yes, you should be ok.

For dialing out, if you use the example showed up, in your sip client, you type 24955, it will be skype echo test.

Did you restart your siptosis after checking the configuration?

If you don't mind, show up in more details on the few configurations file.

The website has the information and some of them can locate in the forum as well.

Unfortunately, the programme development has been terminated. I was left with an old version in my Centos 6 server.

ckleea

The SipOutDialingRules.props should not be empty

You can try my example

1. #rules used by sip caller to make a skype call
   
2. #skypeout dialing transforms:
   
3. #     each rule set on a new line
   
4. #         regex:replacement
   
5. #     if no rule matches, the destination is left unchanged
   
6. #     once a rule is matched, no further processing is done
   
7. 
   
8. #If you are using a PBX and it can't be configured to strip the dialing prefix you can work around the PBX limitation
   
9. #        by adding entries here. If your PBX dialing prefix is 7 you could add the following line:
   
10. #                ^7([1-9][0-9]{10})$:+$1
    
11. #        The $1 will only capture what's in the parenthesis. In the example above, the 7 will be left out when making the SkypeOut call
    
12. #        with 11 digit dialing. If 713051234567 was sent to the this gateway,SipToSis would transform it to +13051234567. For other prefixes, change the 7
    
13. #        to the new prefix. Tip: make sure no other rules make the same match. Read up on regular expressions if you need to get more complicated.
    
14. 
    
15. #You can initiate a conference call to multiple skype users like this (dialing 56 would call the 3 users specified - number of users limited by skype client)
    
16. #You can also do this from a sip device by dialing: skypeuser1,skypeuser2,skypeuser3@SipToSisIpAddress:SipToSisPort
    
17. #^56$:skypeuser1,skypeuser2,skypeuser3
    
18. #you can also make conference to PSTN like this:
    
19. #^57$:+18885555555,+18005555555
    
20. 
    
21. #callback back example to specific users
    
22. #^58$:CallBack:Skype=someskypeuser1,someskypeuser2
    
23. #^58$:CallBack:Skype=someskypeuser1,someskypeuser2|SIP=6666@192.168.0.6:5060
    
24. 
    
25. #callback example with IVR prompt (single target)
    
26. #^58$:CallBack:Skype=someskypeuser1
    
27. #^58$:CallBack:SIP=6666@192.168.0.6:5060
    
28. 
    
29. 
    
30. 
    
31. 
    
32. #rule example below could be for USA and area code 561
    
33. #    dialing 3684111=+15613684111
    
34. #    or 5613684111=+15613684111
    
35. #    or 15613684111=+15613684111
    
36. #    This will also allow international dialing with 7 or more digits.
    
37. #You have to dial with your dial plan prefix if you have one (i.e. #1).
    
38. 
    
39. #^([1-9][0-9]{6})$:+1561$1
    
40. ^([1-9][0-9]{9})$:+1$1
    
41. ^([0-9]{7,})$:+$1
    
42. 
    
43. 
    
44. #you can simulate speed dials this way also (dialing your prefix and 55 would call the skype echo test)
    
45. ^10$:echo123

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To use your sip client, type 410 will get skype echo test

You can add speed dial as

^10$:echo123
^11$:abc
^12$:cde

abc and cde are your skype friends

ckleea

Likely something wrong in the sip to skype dialing connection.
Please post your asterisk log when dialing from sip to skype

ckleea

There is something wrong in your asterisk setup.

Please note that I use Trunkbuilder to set up my siptosis few years ago. The basic codes are the same with only minor settings adjustment

In my sip.conf of asterisk I have

1. [stsTrunk_01]
   
2. username=stsTrunk_01
   
3. type=friend
   
4. secret=yourdefinedsecret
   
5. host=127.0.0.1
   
6. ;nat=no
   
7. dtmfmode=auto
   
8. canreinvite=no
   
9. port=5072
   
10. qualify=yes
    
11. defaultip=127.0.0.1
    
12. incominglimit=1
    
13. outgoinglimit=1
    
14. call-limit=1
    
15. busylevel=1

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The code in my extensions.conf

1. [CallingRule_Skype]; This serve 3 skype trunks _01 _02 _03 _05 _06 _07 _08 & _09 for outgoing to other skype accounts. Only 9 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

stsdialresult is a Macro I have

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. 
   
9. ;ISUP Cause value                        SIP response
   
10. ;  ----------------                        ------------
    
11. ;  1  unallocated number                   404 Not Found
    
12. ;  2  no route to network                  404 Not found
    
13. ;  3  no route to destination              404 Not found
    
14. ;  16 normal call clearing                 --- (*)
    
15. ;  17 user busy                            486 Busy here
    
16. ;  18 no user responding                   408 Request Timeout
    
17. ;  19 no answer from the user              480 Temporarily unavailable
    
18. ;  20 subscriber absent                    480 Temporarily unavailable
    
19. ;  21 call rejected                        403 Forbidden (+)
    
20. ;  22 number changed (w/o diagnostic)      410 Gone
    
21. ;  22 number changed (w/ diagnostic)       301 Moved Permanently
    
22. ;  23 redirection to new destination       410 Gone
    
23. ;  26 non-selected user clearing           404 Not Found (=)
    
24. ;  27 destination out of order             502 Bad Gateway
    
25. ;  28 address incomplete                   484 Address incomplete
    
26. ;  29 facility rejected                    501 Not implemented
    
27. ;  31 normal unspecified                   480 Temporarily unavailable
    
28. exten => s,1,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS})
    
29. exten => s,2(debug1),Verbose(1,debug1 "${HANGUPCAUSE}:${DIALSTATUS}")
    
30. exten => s,3,Set(TIMEOUT(absolute)=120)
    
31. exten => s,4,GotoIf($[${HANGUPCAUSE} = 0]?s,6)
    
32. exten => s,5,Goto(cause-${HANGUPCAUSE},1)
    
33. exten => s,6,GotoIf($[${DIALSTATUS} = NOANSWER]?cause-19,1)
    
34. exten => s,7,GotoIf($[${DIALSTATUS} = BUSY]?cause-2,1)
    
35. exten => s,8,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?cause-0,1)
    
36. exten => s,9,GotoIf($[${DIALSTATUS} = ANSWER]?exit-1,1)
    
37. exten => s,10,Goto(cause-0,1)
    
38. 
    
39. exten => cause-0,1,NoOp(AST_CAUSE_NOTDEFINED)
    
40. exten => cause-0,n,Verbose(1,debug "cause-0")
    
41. exten => cause-0,n,Playback(error)
    
42. ;exten => cause-0,n,Congestion
    
43. exten => cause-0,n,Goto(exit-1,1)
    
44. 
    
45. exten => cause-1,1,NoOp(AST_CAUSE_FAILURE)
    
46. exten => cause-1,n,Verbose(1,debug "cause-1 invalid destination")
    
47. exten => cause-1,n,Playback(invalid)
    
48. exten => cause-1,n,Hangup
    
49. 
    
50. exten => cause-2,1,NoOp(AST_CAUSE_BUSY)
    
51. exten => cause-2,n,Verbose(1,debug "cause-2 busy")
    
52. exten => cause-2,n,Busy
    
53. 
    
54. exten => cause-3,1,NoOp(AST_CAUSE_FAILURE)
    
55. exten => cause-3,n,Verbose(1,debug "cause-3")
    
56. ;exten => cause-3,n,Playback(error)
    
57. exten => cause-3,n,Goto(exit-1,1)
    
58. 
    
59. exten => cause-4,1,NoOp(AST_CAUSE_CONGESTION)
    
60. exten => cause-4,n,Verbose(1,debug "cause-4")
    
61. exten => cause-4,n,Goto(exit-1,1)
    
62. 
    
63. exten => cause-5,1,NoOp(AST_CAUSE_UNALLOCATED)
    
64. exten => cause-5,n,Verbose(1,debug "cause-5 invalid destination")
    
65. exten => cause-5,n,Playback(invalid)
    
66. exten => cause-5,n,Hangup
    
67. 
    
68. exten => cause-18,1,NoOp(AST_CAUSE_CALL_UNPLACED)
    
69. exten => cause-18,n,Verbose(1,debug "cause-18 unplaced")
    
70. exten => cause-18,n,Goto(exit-1,1)
    
71. 
    
72. exten => cause-19,1,NoOp(AST_CAUSE_NO_ANSWER)
    
73. exten => cause-19,n,Verbose(1,debug "cause-19 noanswer")
    
74. exten => cause-19,n,Playback(noanswer)
    
75. exten => cause-19,n,Hangup
    
76. 
    
77. exten => cause-21,1,NoOp(AST_CAUSE_CALL_REJECTED)
    
78. exten => cause-21,n,Verbose(1,debug "cause-21 rejected")
    
79. exten => cause-21,n,Playback(rejected)
    
80. exten => cause-21,n,Hangup
    
81. 
    
82. exten => _cause-X,1,NoOp(UNKNOWN_CAUSECODE)
    
83. exten => _cause-X,n,Verbose(1,debug "cause-X")
    
84. exten => _cause-X,n,Playback(error)
    
85. ;exten => _cause-X,n,Congestion
    
86. exten => _cause-X,n,Goto(exit-1,1)
    
87. 
    
88. exten => exit,1(exit),Noop

複製代碼

Hope this is helpful.

ckleea

Why you need to do asterisk -vvvvc to get asterisk console? Does you asterisk running in the background? What you need to look for the console log is asterisk -vvvr not -vvvvc
I suppect your siptosis is not hooked to the asterisk ie. it is not running in the background.

ckleea

What version linux you are using? Is your asterisk compiled or install via rpm or apt-get?
It seems that your siptosis peers are not configured to take the call. I don't know why from your config files as they all looked ok to me.

For your information, my linux (centos 6) and siptosis work for more than a week from the last reboot. It is running very well.

ckleea

I see. I have not tried siptosis single channel in centos 6.4. There have been some unknown issues in preventing my trunks to start up.

Try centos 5.x, your setup should work.

ckleea

回復 21# kurtor

It should work without any problem.

Follow the instruction again.