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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??

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!

TOP

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
OSSLab Blog :: VoIP & IT Consultant

TOP

回復 21# kurtor

It should work without any problem.

Follow the instruction again.

TOP

I've tried CentOS 5.9 today but it still don't work.
Anyway, thanks for your help

TOP

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.

TOP

I am using Centos 6.4.
Asterisk is download from digium in .tzr.gz format.
I think asterisk is compiled in that case.

TOP

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.

TOP

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.

TOP

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.

TOP

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

  7. #If you are using a PBX and it can't be configured to strip the dialing prefix you can work around the PBX limitation
  8. #        by adding entries here. If your PBX dialing prefix is 7 you could add the following line:
  9. #                ^7([1-9][0-9]{10})$:+$1
  10. #        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
  11. #        with 11 digit dialing. If 713051234567 was sent to the this gateway,SipToSis would transform it to +13051234567. For other prefixes, change the 7
  12. #        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.

  13. #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)
  14. #You can also do this from a sip device by dialing: skypeuser1,skypeuser2,skypeuser3@SipToSisIpAddress:SipToSisPort
  15. #^56$:skypeuser1,skypeuser2,skypeuser3
  16. #you can also make conference to PSTN like this:
  17. #^57$:+18885555555,+18005555555

  18. #callback back example to specific users
  19. #^58$:CallBack:Skype=someskypeuser1,someskypeuser2
  20. #^58$:CallBack:Skype=someskypeuser1,someskypeuser2|SIP=6666@192.168.0.6:5060

  21. #callback example with IVR prompt (single target)
  22. #^58$:CallBack:Skype=someskypeuser1
  23. #^58$:CallBack:SIP=6666@192.168.0.6:5060


  24. #rule example below could be for USA and area code 561
  25. #    dialing 3684111=+15613684111
  26. #    or 5613684111=+15613684111
  27. #    or 15613684111=+15613684111
  28. #    This will also allow international dialing with 7 or more digits.
  29. #You have to dial with your dial plan prefix if you have one (i.e. #1).

  30. #^([1-9][0-9]{6})$:+1561$1
  31. ^([1-9][0-9]{9})$:+1$1
  32. ^([0-9]{7,})$:+$1


  33. #you can simulate speed dials this way also (dialing your prefix and 55 would call the skype echo test)
  34. ^50$:echo123
  35. ^57$:echo123
  36. ^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

TOP

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

  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
Hope this is helpful.

TOP

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'

TOP

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

TOP

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

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