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標題: Problem in SipToSis [打印本頁]
作者: kurtor    時間: 2013-4-21 18:58     標題: Problem in SipToSis

SipToSis can receive incoming call from Skype but it cannot make call to skype user.
I've followed the procedures in http://www.telecom-cafe.com/forum/viewthread.php?tid=3682 and I go to Setup SipToSis with a SIP Phone/ATA - Configuration and follow those procedures.
I was stucked in *,*,localnet,calleeid (default) this line.
I don't know whether I should type the IP of Asterisk server and calleeid like (echo123) in that line or just type the previous line onto SipToSkypeAuth.props.
I cannot make call to asterisk user even if I follow the setting, I don't know what's the reason behind.
Could you guys please help?
Is there any other reference or tutorial guiding users to setup SipToSis??
作者: alang    時間: 2013-4-22 11:33

You don't need to make any changes to the SipToSkypeAuth.props, just to keep it on the default settings.

Please be sure the line below existed in the SkypeOutDialingRules.props and the destination number is 55.

^55$:echo123
作者: kurtor    時間: 2013-4-25 10:03

1. I've made the following setting but it still failed to call from Asterisk server to Skype user.
2. In SkypeToSipAuth.props, Should I use Asterisk server's IP address and port number or the called party's IP address and port number?
3. When user terminates SkypeToSis call in VoIP, call was not terminated in Skype side. What's the cause of that?
P.S. SIP user can receives call from Skype if the following configuration is used. Skype, SipToSis and Asterisk were installed onto the same machine

Here is the configuration file used in SipToSis:

In extensions.conf:
[LocalSets]
...
exten => _249.,1,Verbose(2,SkypeOut)
same => n,Answer()
same => n,Dial(SIP/skypetestuser/{EXTEN:3})

In sip.conf:
[skypetestuser]
username=skypetestuser
type=friend
context=LocalSets
secret=skypetest
host=dynamic
nat=auto_force_rport
dtmfmode=auto
canreinvite=no
qualify=yes
incominglimit=1
outgoinglimit=1
call=limit=1
busylevel=1

In SkypeOutDialingRules.props:
^55$:echo123

In SipToSkypeAuth.props:
*,*,localnet,calleeid

In siptosis.cfg:
#Sample Asterisk registration example - comment out NO registration info above first and uncomment the following.
host_port=5070
contact_url=sip:skypetestuser@127.0.0.1:5070
from_url="skypetestuser" <sip:skypetestuser@127.0.0.1:5060>
username=skypetestuser
realm=asterisk
passwd=skypetest
expires=3600
do_register=yes
minregrenewtime=120
regfailretrytime=15
#Puts your fake (or real) DID in RequestLine
DIDNumber=15551234567
#Puts your fake (or real) DID in To Header
DIDNumber=to:15551234567
# --- end of Asterisk Reg example ---

And In SkypeToSipAuth.props:
*,sip:bob@192.168.200.10:53337   
# bob is a sip user defined in the server

Many Thanks!!!!!!
作者: 角色    時間: 2013-4-25 22:33

Have you read the previous threads that were written by other members in this forum?

Based on my past experience (Installed SiptoSis on Windows Xp and CentOS Linux environments), you have to read the configuration files in SiptoSis thoroughly since there is plenty of information for you to adjust/set/modified in accordance with your desired operation.
作者: ckleea    時間: 2013-4-26 06:43

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.
作者: kurtor    時間: 2013-4-26 12:17

回復 4# 角色

My plan is to install SipToSis and Asterisk on the same computer using CentOS.
I tried to following the instructions in  
http://www.telecom-cafe.com/forum/viewthread.php?tid=3682 but I cannot make it.
it is probably because I missed something but I don't know which part I've missed.

Based on your experience, can you name some parts that I may set wrongly and lead to this problem?

Thanks
作者: kurtor    時間: 2013-4-26 12:18

回復 5# ckleea

In Asterisk CLI, after executing sip show peers, I found siptosis client online.
Thus, I can receive call from skype.
But i cannot make call from VoIP to Skype.

Thanks
作者: ckleea    時間: 2013-4-26 15:30

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.
作者: kurtor    時間: 2013-4-27 01:23

回復 8# ckleea

In SkypeOutDialingRules.props:
^55$:echo123
# only this line is used.


In extensions.conf:
exten => _4.,1,Dial(SIP/skypetestuser/${EXTEN:1})


In SkypeToSipAuth.props:
*,sip:kurt@192.168.200.10:53337
# only this line is used.


In SipToSkypeAuth.props:
*,*,localnet,calleeid
# only this line is used.


In SipOutDialingRules.props:
# It is an empty file.


In stsTrunk_linux:
#Setup environment here if needed
#export JAVA_HOME=/opt/java
#export PATH=$PATH:/opt/java:/usr/bin
export PATH=$PATH:/sbin

#User to run skype and siptosis under
runuser=kurt

#path where siptosis and this script are located - manually set if needed
fullpath="$(cd "${0%/*}" 2>/dev/null; echo "$PWD"/"${0##*/}")"
scriptpath=`dirname "${fullpath}"`
#scriptpath=/home/kurt/siptosis

#*** some config settings ***

#if want to use vnc, set to N - vncserver launches it by default
usetwm=N
#set to Y if using twm or vnc with twm
testtwmconfig=N
#if using twm or vnc with twm, edit /etc/X11/twm/system.twmrc and add 'RandomPlacement' above 'NoGrabServer' line
twmrcfile=/etc/X11/twm/system.twmrc

#can be set to XVFB, VNC or NORMAL (uppercase only) - if using vnc, run vncpasswd to set the password
displayMethod=XVFB

#set for XVFB and VNC displayMethod
displaynum=101

#*** end of config settings ***

siptosis.cfg is attached here
作者: ckleea    時間: 2013-4-27 07:14

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

  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.

  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

  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

  25. #callback example with IVR prompt (single target)
  26. #^58$:CallBack:Skype=someskypeuser1
  27. #^58$:CallBack:SIP=6666@192.168.0.6:5060




  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).

  39. #^([1-9][0-9]{6})$:+1561$1
  40. ^([1-9][0-9]{9})$:+1$1
  41. ^([0-9]{7,})$:+$1


  44. #you can simulate speed dials this way also (dialing your prefix and 55 would call the skype echo test)
  45. ^10$:echo123
複製代碼
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
作者: kurtor    時間: 2013-4-27 09:34

本帖最後由 kurtor 於 2013-4-27 09:38 編輯

回復 10# ckleea

After using your code, i still cannot dial to Skype Test Call by extension 410.

I've tried to update SkypeOutDialingRules.props and then restart the computer.
After that, I start skype first. Then I execute ./SipToSis_linux in the terminal.

The Error code from VoIP user is 503 Service Unavailable.
I don't know the cause of that.

The output of SipToSis_linux is shown below:
Launching SipToSis
### Warning: Duplicate Parameter: didnumber
2013-04-27 01:21:57,393 Starting SipToSis v20111012
2013-04-27 01:21:57,438 Skype4Java Version 1.3.0.1
2013-04-27 01:21:57,438 os=Linux ver=2.6.32-358.6.1.el6.i686 arch=i386 (4 core)
2013-04-27 01:21:57,438 javaVer=1.7.0_19 - Oracle Corporation (32 bit)
2013-04-27 01:21:57,467 Available Codecs: PCMU(0),PCMA(8),iLBC(98),L16/16k(102)
2013-04-27 01:21:57,467 DTMF rfc2833(101)
2013-04-27 01:21:57,475 initSkype - If stuck, check Skype online & API auth
2013-04-27 01:22:00,664 SkypeVer:175
2013-04-27 01:22:00,740 Attached SkypeUserId:kurt.or
2013-04-27 01:22:00,766 Config - skypeClientSupportsMultiCalls:false  concurrentCallLimit:2
2013-04-27 01:22:00,766 SipToSis contact_url=sip:skypetestuser@127.0.0.1:5070
2013-04-27 01:22:00,766 Ext. Srvr (from_url)=sip:skypetestuser@127.0.0.1:5060
2013-04-27 01:22:00,766 via_addr=192.168.200.11  realm=asterisk
2013-04-27 01:22:00,766 RTP Ports: 63200-63202  Local Skype Ports: 64432-64435
2013-04-27 01:22:00,766 jitterLevel=-1
2013-04-27 01:22:00,832 Registrar Server Domains=
2013-04-27 01:22:00,839 WAITING FOR INCOMING CALL
2013-04-27 01:22:00,839 WAITING FOR INCOMING CALL
2013-04-27 01:22:00,859 MaxCallTime: not limited MaxPSTNCallTime: not limited
2013-04-27 01:22:00,859 MaxDailyPSTNUniqueNumberCount: 48 MaxDailyPSTNMinutes: 350
2013-04-27 01:22:00,860 Loading Skype PSTN Call History
2013-04-27 01:22:00,890 0 possible calls to import.
2013-04-27 01:22:00,930 PSTN counters reset at: 00:00:00 GMT
2013-04-27 01:22:00,932 Qualified PSTN calls today: 0 Time: 0 minutes
2013-04-27 01:22:01,041 AcctBalance: 0.00
2013-04-27 01:22:01,041 REGISTRATION
2013-04-27 01:22:08,077 STUN: No Response.
2013-04-27 01:22:08,581 PublicIP=119.236.2.54
2013-04-27 01:22:12,569 401 Registration Authentication
2013-04-27 01:22:12,580 Registration success: 200 OK
作者: ckleea    時間: 2013-4-27 21:23

Likely something wrong in the sip to skype dialing connection.
Please post your asterisk log when dialing from sip to skype
作者: kurtor    時間: 2013-4-27 23:49

本帖最後由 kurtor 於 2013-4-27 23:51 編輯

回復 12# ckleea

Here's the log in Asterisk CLI using asterisk -vvvvc:

SecurityEvent="ChallengeSent",EventTV="1367077003-192377",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:kurt@192.168.200.11",SessionID="0xb765afe4",LocalAddress="IPV4/UDP/192.168.200.11/5060",RemoteAddress="IPV4/UDP/192.168.200.10/53337",Challenge="7b80c0e1"
  == Using SIP RTP CoS mark 5
    -- Executing [410@LocalSets:1] Dial("SIP/kurt-00000002", "SIP/skypetestuser/10") in new stack
Agent policy for SIP/kurt-00000002 is 'never'. CC not possible
  == Using SIP RTP CoS mark 5
Don't know any of (nothing) formats
No audio format found to offer. Cancelling call to 10
    -- Couldn't call SIP/skypetestuser/10
  == Everyone is busy/congested at this time (0:0/0/0)
    -- Auto fallthrough, channel 'SIP/kurt-00000002' status is 'CHANUNAVAIL'
作者: ckleea    時間: 2013-4-28 06:41

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
複製代碼
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

  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)

  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)

  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

  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

  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)

  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)

  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

  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)

  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

  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

  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)

  88. exten => exit,1(exit),Noop
複製代碼
Hope this is helpful.
作者: kurtor    時間: 2013-4-28 11:57

回復 14# ckleea
I've tried the setting and Asterisk returns error messages.
I've also attached the setting files for reference.

in Asterisk -vvvvc
  1. == Using SIP RTP CoS mark 5
  2.     -- Executing [450@LocalSets:1] NoOp("SIP/bob-00000000", "") in new stack
  3.     -- Executing [450@LocalSets:2] Dial("SIP/bob-00000000", "SIP/skypetestuser/50") in new stack
  4.   == Using SIP RTP CoS mark 5
  5.     -- Couldn't call SIP/skypetestuser/50
  6.   == Everyone is busy/congested at this time (0:0/0/0)
  7.     -- Executing [450@LocalSets:3] Macro("SIP/bob-00000000", "stsdialresult") in new stack
  8.     -- Executing [s@macro-stsdialresult:1] NoOp("SIP/bob-00000000", "HANGUPCAUSE is 0 and DIALSTATUS is CHANUNAVAIL") in new stack
  9.     -- Executing [s@macro-stsdialresult:2] Verbose("SIP/bob-00000000", "1,debug1 "0:CHANUNAVAIL"") in new stack
  10. debug1 "0:CHANUNAVAIL"
  11.     -- Executing [s@macro-stsdialresult:3] Set("SIP/bob-00000000", "TIMEOUT(absolute)=120") in new stack
  12.     -- Channel will hangup at 2013-04-28 11:40:07.975 HKT.
  13.     -- Executing [s@macro-stsdialresult:4] GotoIf("SIP/bob-00000000", "1?s,6") in new stack
  14.     -- Goto (macro-stsdialresult,s,6)
  15.     -- Executing [s@macro-stsdialresult:6] GotoIf("SIP/bob-00000000", "0?cause-19,1") in new stack
  16.     -- Executing [s@macro-stsdialresult:7] GotoIf("SIP/bob-00000000", "0?cause-2,1") in new stack
  17.     -- Executing [s@macro-stsdialresult:8] GotoIf("SIP/bob-00000000", "1?cause-0,1") in new stack
  18.     -- Goto (macro-stsdialresult,cause-0,1)
  19.     -- Executing [cause-0@macro-stsdialresult:1] NoOp("SIP/bob-00000000", "AST_CAUSE_NOTDEFINED") in new stack
  20.     -- Executing [cause-0@macro-stsdialresult:2] Verbose("SIP/bob-00000000", "1,debug "cause-0"") in new stack
  21. debug "cause-0"
  22.     -- Executing [cause-0@macro-stsdialresult:3] Playback("SIP/bob-00000000", "error") in new stack
  23.        > 0xb750f960 -- Probation passed - setting RTP source address to 192.168.200.10:4010
  24.     -- Executing [cause-0@macro-stsdialresult:4] Goto("SIP/bob-00000000", "exit-1,1") in new stack
  25.     -- Goto (macro-stsdialresult,exit-1,1)
  26.     -- Executing [450@LocalSets:4] Playback("SIP/bob-00000000", "pls-try-call-later") in new stack
  27.     -- Auto fallthrough, channel 'SIP/bob-00000000' status is 'CHANUNAVAIL'
複製代碼
In extensions.conf, I've added
  1. exten => _4.,1,Noop
  2. exten => _4.,n,Dial(SIP/skypetestuser/${EXTEN:1})
  3. exten => _4.,n,Macro(stsdialresult)
  4. exten => _4.,n,Playback(pls-try-call-later)
複製代碼
in sip.conf
  1. [skypetestuser]        ;SipToSis
  2. username=skypetestuser
  3. type=friend
  4. secret=skypetest
  5. host=dynamic
  6. nat=no
  7. dtmfmode=auto
  8. canreinvite=no
  9. port=5074
  10. qualify=yes
  11. defaultip=192.168.200.11
  12. incominglimit=1
  13. outgoinglimit=1
  14. call-limit=1
  15. busylevel=1
複製代碼
SkypeOutDialingRules.props:
  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

  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.

  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

  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

  25. #callback example with IVR prompt (single target)
  26. #^58$:CallBack:Skype=someskypeuser1
  27. #^58$:CallBack:SIP=6666@192.168.0.6:5060


  30. #rule example below could be for USA and area code 561
  31. #    dialing 3684111=+15613684111
  32. #    or 5613684111=+15613684111
  33. #    or 15613684111=+15613684111
  34. #    This will also allow international dialing with 7 or more digits.
  35. #You have to dial with your dial plan prefix if you have one (i.e. #1).

  37. #^([1-9][0-9]{6})$:+1561$1
  38. ^([1-9][0-9]{9})$:+1$1
  39. ^([0-9]{7,})$:+$1


  42. #you can simulate speed dials this way also (dialing your prefix and 55 would call the skype echo test)
  43. ^50$:echo123
  44. ^57$:echo123
  45. ^58$:Skype=echo123
複製代碼
siptosis.cfg, generated by stsTrunkBuilder
  1. #stsTrunkBuilder 20110808 generated trunk file 01 created: Sun Apr 28 01:04:17 GMT 2013 3120073164
  2. logConfigFile=log.properties
  3. recordSkypeIn=no
  4. recordSkypeOut=no
  5. recordSIPIn=no
  6. recordSIPOut=no
  7. runConnectorReliabilityTest=no
  8. linux_lockmode=0
  9. linux_idleSleepMs=150
  10. linux_activeSleepMs=75
  11. skypeAPITrace=no
  12. configWatchInterval=0
  13. AuthWaitingPollIntervalSeconds=0
  14. connectorWatchDogMinutes=0
  15. connectionFee=0
  16. MaxCallTimeLimitMinutes=0
  17. WarnMinutesBeforeCutoff=1
  18. OverLimitWarningFile=clips/overlimit.wav
  19. OverUsageLimitSipResponse=480
  20. dailyPstnLimitMinutes=350
  21. dailyPstnUniqueNumberLimit=48
  22. refuseNewPstnCallsWhenRemainingMinutesUnder=5
  23. MaxPstnCallTimeLimitMinutes=0
  24. loadSkypeClientCallHistory=yes
  25. tollFreeNumberPrefixes=1800,1888,1866,1877
  26. autoStartVideoMakeCall=no
  27. autoStartVideoAnswerCall=no
  28. emailWhenBalanceDropsTo=-1
  29. emailHost=
  30. emailPort=25
  31. emailusername=skypetestuser
  32. emailPassword=
  33. emailRecipients=
  34. emailFrom=
  35. emailTest=no
  36. setSkypeOnlineStatusInterval=0
  37. skypeOnlineStatus=ONLINE
  38. callBackForceSipPrefix=*
  39. callLogPath=log/
  40. siptoskypeauthfile=SipToSkypeAuth.props
  41. skypetosipauthfile=SkypeToSipAuth.props
  42. SkypeOutDialingRulesFile=SkypeOutDialingRules.props
  43. SipOutDialingRulesFile=SipOutDialingRules.props
  44. ua_jar=ua.jar
  45. audioPriorityIncrease=0
  46. jitterLevel=-1
  47. skype_connect=yes
  48. skype_audioportbase=64438
  49. enableSkypeDtmfDetector=yes
  50. SkypeDtmfGain=1
  51. SkypeDtmfDetectorHitThreshold=30
  52. SkypeDtmfDetectorSilenceThreshold=40
  53. SkypeDtmfDetectorTwistAdjust=1
  54. autoDisableSkypeDtmfDetectorSeconds=80
  55. sendSipDtmfToSkype=yes
  56. sendSkypeDtmfToSip=yes
  57. skypeDtmfInterDigitDelayms=700
  58. inbandFullTimeDtmfDetection=yes
  59. suppressSkypePSTNSipRingback=no
  60. JoinManualSkypeOutboundCallToSip=no
  61. SkypeInboundAllChannelsBusyAction=ignore
  62. SkypeTransferTimeoutMs=8000
  63. SkypeInboundSipDestUnavailableAction=refuse
  64. SipInboundAllChannelsBusyAction=busy
  65. skypeclientsupportsmulticalls=no
  66. concurrentcalllimit=1
  67. autoShutdownMinutes=0
  68. pintimeout=8
  69. pinretrylimit=3
  70. destinationtimeout=12
  71. destinationretrylimit=3
  72. pinFile=clips/enterPin.wav
  73. destinationFile=clips/enterDest.wav
  74. dialingFile=clips/dialing.wav
  75. invalidPinFile=clips/invalidPin.wav
  76. invalidDestFile=clips/invalidDest.wav
  77. skypePinFile=clips/enterPin.wav
  78. skypeDestinationFile=clips/enterDest.wav
  79. skypeDialingFile=clips/dialing.wav
  80. skypeInvalidPinFile=clips/invalidPin.wav
  81. skypeInvalidDestFile=clips/invalidDest.wav
  82. handleEarlyMedia=yes
  83. handleSipEarlyMedia=no
  84. sendRingToSkypeCaller=no
  85. skypeRingFile=clips/skypeRing.wav
  86. skypeRingInterval=8
  87. sendSkypeEarlyMediaOverSipSessionProgress=no
  88. replaceFromWithSkypeId=no
  89. sendSkypeIM=no
  90. skypeimmessage=You are about to receive a Skype Voice call from [callerid] [callernumber].
  91. sendSkypeImDelay=2
  92. autoAddContactCalledUsers=no
  93. autoAddContactAuthMessage=Please authorize me to contact you.
  94. transport_protocols=udp
  95. stunServer=stun.xten.net:3478,stunserver.org:3478
  96. stunTestInterval=30
  97. enableNatTranslate=yes
  98. enableClineFix=yes
  99. enableNatTranslateVia=no
  100. host_port=5074
  101. from_url="skypetestuser" <sip:skypetestuser@192.168.200.11:5060>
  102. username=skypetestuser
  103. realm=asterisk
  104. passwd=skypetest
  105. expires=3600
  106. do_register=yes
  107. minregrenewtime=120
  108. regfailretrytime=15
  109. DIDNumber=15551234567
  110. DIDNumber=to:15551234567
  111. keepalive_time=0
  112. audio=yes
  113. audio_port=63204
  114. noRtpReceivedAutoHangupSeconds=30
  115. audio_codec=PCMU,PCMA,ILBC,L16_16
  116. audio_frame_size=240,240,240,160
  117. audio_avp=-1,-1,98,102
  118. skype_audiooutgain=1,1,1,1
  119. skype_audioingain=1.2,1.2,1.2,1
  120. FilterParams=NONE
  121. enableSendRTPtoReceivedAddress=yes
  122. lockRtpSendAddressAfterPackets=1
  123. dtmf2833payloadtype=101
  124. enableSIPInbandDtmfDetector=no
  125. SIPInbandDtmfDetectorGain=1
  126. SipDtmfDetectorHitThreshold=30
  127. SipDtmfDetectorSilenceThreshold=40
  128. SipDtmfDetectorTwistAdjust=1
  129. autoDisableSipInbandDtmfDetectorSeconds=80
  130. useViaRport=yes
  131. useViaReceived=yes
  132. enableFixRemoteAddress=yes
  133. sendResponseUsingOutboundProxy=yes
  134. early_dialog=yes
  135. baseFailureResponse=403
  136. skypeRefusedResponse=603
  137. skypeFailedResponse=408
  138. skypeInvalidDestinationResponse=404
  139. skypeUnPlacedResponse=408
  140. skypeBusyResponse=600
  141. SkypeUserOfflineFailureResponse=410
  142. TcpRxBufferSize=8192
  143. TcpTxBufferSize=8192
  144. RtpRxBufferSize=8192
  145. RtpTxBufferSize=8192
  146. RtpTosFlags=0x10
  147. is_registrar=no
  148. register_new_users=yes
  149. allowMultiContactsPerUser=no
  150. domain_port_any=yes
  151. vmNoMessagesClip=clips/vmpnomessages.wav
  152. vmPlayingClip=clips/vmpplayingmessages.wav
  153. vmEndOfMessageClip=clips/vmpendofmessageprompt.wav
  154. vmMessageDeletedClip=clips/vmpmessagedeleted.wav
  155. vmMessageDeleteAllClip=clips/vmpmessagedeleteall.wav
  156. vmNoMoreMessagesClip=clips/vmpnomoremessages.wav
  157. vmRecordingBeepClip=clips/vmprecordingbeep.wav
複製代碼
作者: ckleea    時間: 2013-4-28 14:44

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.
作者: kurtor    時間: 2013-4-29 01:18

I use asterisk -vvvvc because asterisk is not started after I logon to the computer.
I just don't know how to check siptosis is hooked to asterisk.
It seems that siptosis is registered onto the Asterisk server correctly as I can see my siptosis account online when I type sip show peers in Asterisk CLI.
作者: ckleea    時間: 2013-4-29 06:50

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.
作者: kurtor    時間: 2013-4-29 09:36

I am using Centos 6.4.
Asterisk is download from digium in .tzr.gz format.
I think asterisk is compiled in that case.
作者: ckleea    時間: 2013-4-29 13:59

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.
作者: kurtor    時間: 2013-5-2 00:20

I've tried CentOS 5.9 today but it still don't work.
Anyway, thanks for your help
作者: ckleea    時間: 2013-5-2 05:45

回復 21# kurtor

It should work without any problem.

Follow the instruction again.
作者: alang    時間: 2013-5-2 13:48

I use asterisk -vvvvc because asterisk is not started after I logon to the computer.
I just don't kn ...
kurtor 發表於 2013-4-29 01:18


kurtor,
If you still didn't get it to work I really suggest you doing the setup from basis configuration, that means don't use the main configuration file 'siptosis.cfg' generated by stsTrunkBuilder unless you've realized every variables within the file.

Here are basis configuration for siptosis.cfg

host_port=5070
username=skypests
passwd=unimportantpassword
do_register=no

The proper trunk setting on asterisk/FreePBX

type=peer
host=<SipToSis LAN IP>
port=5070
nat=no
dtmfmode=auto
canreinvite=no
qualify=yes
incominglimit=1
outgoinglimit=1
call-limit=1
busylevel=1
context=from-trunk

Note: Be sure you've closed the SELinux & firewall on CentOS.

Good luck
作者: 浮雲1965    時間: 2013-6-5 20:12

我的情況同樓主一樣。我是使用Elastix， CentOS5.4
SKype可以打入Elastix，但Elastix的分機不能打出Skype。
請成功的朋友提供你們的siptosis的各個配置文件sample.

Thanks！



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