ckleea

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

ckleea

回復 44# 角色

There are four config files need to set.

1. ststrunk.cfg
- need to check the port number and config files name for each trunk

2. siptoskypeauth.props
no change

3. skypetosipauth.props.
-define your incoming skype call to which asterisk extensions

4. skyeoutdialingrules.props.
-define your speed dial code

ckleea

Please show me the file names for the one channel version

ckleea

http://www.mhspot.com/sts/siptosis_skype_trunk_howto.html

Please follow also the the above link to install the trunk.

ckleea

回復 49# bubblestar


    What is the name of your second file? It is not the same as mine. My stsTrunk_01.cfg is much longer

ckleea

回復 51# bubblestar
this is the nomenclature I used for multiple trunks. Please send me the files for a look

ckleea

回復 53# bubblestar


See my version

Please note that I use _0X to denote individual trunk

1. #stsTrunkBuilder 20091019 generated trunk file 01 created: Tue Feb 15 17:04:58 HKT 2011 1527885965
   
2. logConfigFile=stsTrunk_01_log.properties
   
3. runConnectorReliabilityTest=no
   
4. skypeAPITrace=no
   
5. configWatchInterval=0
   
6. connectorWatchDogMinutes=0
   
7. connectionFee=0
   
8. MaxCallTimeLimitMinutes=0
   
9. WarnMinutesBeforeCutoff=1
   
10. OverLimitWarningFile=clips/overlimit.wav
    
11. OverUsageLimitSipResponse=480
    
12. dailyPstnLimitMinutes=350
    
13. dailyPstnUniqueNumberLimit=48
    
14. refuseNewPstnCallsWhenRemainingMinutesUnder=5
    
15. MaxPstnCallTimeLimitMinutes=0
    
16. loadSkypeClientCallHistory=yes
    
17. tollFreeNumberPrefixes=1800,1888,1866,1877
    
18. emailWhenBalanceDropsTo=-1
    
19. emailHost=
    
20. emailPort=25
    
21. emailusername=stsTrunk_01
    
22. emailPassword=
    
23. emailRecipients=
    
24. emailFrom=
    
25. emailTest=no
    
26. setSkypeOnlineStatusInterval=0
    
27. skypeOnlineStatus=ONLINE
    
28. callBackForceSipPrefix=*
    
29. callLogPath=log/
    
30. siptoskypeauthfile=SipToSkypeAuth_01.props
    
31. skypetosipauthfile=SkypeToSipAuth_01.props
    
32. SkypeOutDialingRulesFile=SkypeOutDialingRules_01.props
    
33. SipOutDialingRulesFile=SipOutDialingRules_01.props
    
34. ua_jar=ua.jar
    
35. audioPriorityIncrease=0
    
36. jitterLevel=2
    
37. skype_connect=yes
    
38. skype_audioportbase=64436
    
39. enableSkypeDtmfDetector=yes
    
40. SkypeDtmfDetectorHitThreshold=40
    
41. SkypeDtmfDetectorSilenceThreshold=6
    
42. sendSipDtmfToSkype=yes
    
43. sendSkypeDtmfToSip=yes
    
44. inbandFullTimeDtmfDetection=yes
    
45. JoinManualSkypeOutboundCallToSip=no
    
46. SkypeInboundAllChannelsBusyAction=transferto:[color=Red]your_skype_username[/color]
    
47. SkypeTransferTimeoutMs=8000
    
48. SkypeInboundSipDestUnavailableAction=refuse
    
49. SipInboundAllChannelsBusyAction=busy
    
50. skypeclientsupportsmulticalls=no
    
51. concurrentcalllimit=1
    
52. autoShutdownMinutes=0
    
53. pintimeout=8
    
54. pinretrylimit=3
    
55. destinationtimeout=12
    
56. destinationretrylimit=3
    
57. pinFile=clips/enterPin.wav
    
58. destinationFile=clips/enterDest.wav
    
59. dialingFile=clips/dialing.wav
    
60. invalidPinFile=clips/invalidPin.wav
    
61. invalidDestFile=clips/invalidDest.wav
    
62. skypePinFile=clips/enterPin.wav
    
63. skypeDestinationFile=clips/enterDest.wav
    
64. skypeDialingFile=clips/dialing.wav
    
65. skypeInvalidPinFile=clips/invalidPin.wav
    
66. skypeInvalidDestFile=clips/invalidDest.wav
    
67. handleEarlyMedia=yes
    
68. handleSipEarlyMedia=no
    
69. sendRingToSkypeCaller=no
    
70. skypeRingFile=clips/skypeRing.wav
    
71. skypeRingInterval=8
    
72. sendSkypeEarlyMediaOverSipSessionProgress=yes
    
73. replaceFromWithSkypeId=no
    
74. sendSkypeIM=no
    
75. skypeimmessage=You are about to receive a Skype Voice call from [callerid] [callernumber].
    
76. sendSkypeImDelay=2
    
77. transport_protocols=udp
    
78. stunTestInterval=30
    
79. enableNatTranslate=yes
    
80. enableNatTranslateVia=no
    
81. host_port=5072
    
82. username=stsTrunk_01
    
83. passwd=your_generated_passkey
    
84. do_register=no
    
85. keepalive_time=0
    
86. audio=yes
    
87. audio_port=63204
    
88. noRtpReceivedAutoHangupSeconds=30
    
89. audio_codec=PCMU,PCMA,ILBC
    
90. audio_frame_size=240,240,240
    
91. audio_avp=-1,-1,98
    
92. skype_audiooutgain=1,1,1
    
93. skype_audioingain=1.5,1.5,1.5
    
94. FilterParams=NONE
    
95. enableSendRTPtoReceivedAddress=yes
    
96. lockRtpSendAddressAfterPackets=1
    
97. dtmf2833payloadtype=101
    
98. enableSIPInbandDtmfDetector=no
    
99. SipDtmfDetectorHitThreshold=30
    
100. SipDtmfDetectorSilenceThreshold=6
     
101. useViaRport=yes
     
102. useViaReceived=yes
     
103. sendResponseUsingOutboundProxy=no
     
104. baseFailureResponse=403
     
105. skypeRefusedResponse=603
     
106. skypeFailedResponse=404
     
107. skypeUnPlacedResponse=408
     
108. skypeBusyResponse=600
     
109. TcpRxBufferSize=8192
     
110. TcpTxBufferSize=8192
     
111. RtpRxBufferSize=8192
     
112. RtpTxBufferSize=8192
     
113. is_registrar=yes
     
114. register_new_users=yes
     
115. allowMultiContactsPerUser=no

複製代碼

ckleea

回復 55# bubblestar
大部分都是跟 default，慢慢試。

ckleea

回復 57# bubblestar


    Somethings must be changed

ckleea

Please also note

Call-Back Setup Instructions

Note: PSTN rates will be per PSTN outbound call according to the selected provider billing terms.

Single Stage Callback - no IVR or additional dialing - Scroll down for two stage metbod.

Trigger callback using a Skype DID (AKA SkypeIn,Skype Online number) or from a Specific Skype User
Edit SkypeToSipAuth.props
Add a line like this (at least two targets):
AuthorizedSkypeIdOrNumber,CallBack:Skype=someid1OrPstnNumber,someskypeid2OrPstnNumber
or:
AuthorizedSkypeIdOrNumber,CallBack:Skype=someid1OrPstnNumber|SIP=someSipAddress@someprovider:5060
Note: only a single SIP target can be specified.


Using a SIP DID to trigger callback
Edit SipToSkypeAuth.props
Add a line like this (at least two targets):
AuthorizedSIPNumber,*,*,CallBack:Skype=someSkypeId1OrPstnNumber,someSkypeId2OrPstnNumber
or:
AuthorizedSIPNumber,*,*,CallBack:Skype=someSkypeId1,someSkypeId2|SIP=someSipAddress@someprovider:5060
Note: only a single SIP target can be specified.



Another way of using a SIP call to trigger callback
Edit SkypeOutDialingRules.props
Add a line like this (at least two targets):
^58$:CallBack:Skype=someskypeuser1OrPstnNumber,someskypeuser2OrPstnNumber
or:
^58$:CallBack:Skype=someskypeuser1OrPstnNumber,someskypeuser2OrPstnNumber|SIP=someSIPUser@SomeSIPAddress:5060
In this example, if you dial 58@yourSTSGateway:stsPort - it will trigger a callback.
Note: only a single SIP target can be specified.


Two Stage Callback - uses IVR for dialing
Note: DTMF decoding must be on. In the case of a Skype PSTN target, DTMF decoding may not be reliable.

Trigger callback using a Skype DID (AKA SkypeIn,Skype Online number) or from a Specific Skype User
Edit SkypeToSipAuth.props
Add a line like this (specify only one target):
AuthorizedSkypeIdOrNumber,CallBack:Skype=someSkypeIdOrPSTNNumber
or:
AuthorizedSkypeIdOrNumber,CallBack:SIP=someSipAddress@someprovider:5060
Once the called back target answers, the IVR will prompt for the destination.
Destination will be dialed as defined in SkypeOutDialingRules.props.
Default is to call out via Skype, to dial out using SIP instead, dial * before the destination.
Parameter callBackForceSipPrefix controls the SIP dialing prefix.
In the case of SIP dialing, destination will be dialed as defined in SipOutDialingRules.

Using a SIP DID to trigger callback
Edit SipToSkypeAuth.props
Add a line like this (specify only one target):
AuthorizedSIPNumber,*,*,CallBack:Skype=someSkypeIdOrPSTNNumber
or:
AuthorizedSIPNumber,*,*,CallBack:SIP=someSipAddress@someprovider:5060
Once the called back target answers, the IVR will prompt for the destination.
Destination will be dialed as defined in SkypeOutDialingRules.props.

Another way of using a SIP call to trigger callback
Edit SkypeOutDialingRules.props
Add a line like this (specify only one target):
^58$:CallBack:Skype=someSkypeIdOrPSTNNumber
or:
^58$:CallBack:SIP=someSIPUser@SomeSIPAddress:5060
In this example, if you dial 58@yourSTSGateway:stsPort - it will trigger a callback.
Once the called back target answers, the IVR will prompt for the destination.
Destination will be dialed as defined in SkypeOutDialingRules.props.

ckleea

To enable all on status, go to the config.xml and change

      <IdleTimeForAway>0</IdleTimeForAway>
      <IdleTimeForNA>0</IdleTimeForNA>

Then your skype status won't be away

ckleea

For your information, i believe the paid version of stsTrunkbuilder has been discontinued and withdrawn from usage.

Though i download the free version from the site, the content is completely different from paid version that i have.

ckleea

回復 65# 角色


I look at the props and cfg files. There have been more settings in the new version.
I am unable to replace my old files with the new one as it would overwrite mine. What I did before, use trunkbuilder to build the trunk settings, then copy to sip.conf. Config each skype instance to accept skype-java and auto login etc.

Then autoboot with /etc/rc.d/rc.local

ckleea

Siptosis has a lot of settings needed to look at. It seems very unfortunate that the developer has taken out the paid version from the site and will provide limited support only for paid users.
I am looking forward for an upgrade.

ckleea

I have just tried this combination

skype in US -> siptosis on asterisk in HK -> sip phone in UK

Almost like the usual copper wire analog phone calls.