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SippySkype Skype/Sip Gateway V3 20080725 Update

Latest changes: V3 - limited testing
  • Fixed call logging
  • Fixed dialing rules file case
  • New param logConfigFile
  • RTP enhancements


Download at http://www.mhspot.com/mhspot/sippyskype.htm as usual.

SippySkype V3 20080722 - New version

First version of v3 - limited testing

      ---- You must add some new parameters if upgrading from v2----

  • Refactored to handle multiple call streams - Win Skype Clients 3.6 - 3.8 do not support multi active calls - do any? A non supporting skype client will show on console "A call exists in INPROGRESS" if attempting multiple outbound calls.
  • Changed linux startup script to linux/unix format
  • Added exit code so scripts can detect startup failure.
  • Now supports SIP Hold
  • Now supports Skype Hold
  • Changed to skype4java version as of 2008/07/14
  • Fix EarlyMedia with pin issue
  • Fix 401/407 INVITE Cancel problem
  • Parameters skype_audioinport and skype_audiooutport merged to new single parameter skype_audioportbase
  • New parameter concurrentcalllimit
  • New parameter skypeclientsupportsmulticalls
  • New parameter SkypeOutDialingRules - set your area dialing preferences - look at cfg file for example
  • New parameter sendSipDtmfToSkype
  • New parameter sendSkypeDtmfToSip - not working
  • New parameter SkypeInboundAllChannelsBusyAction

     Download at http://www.mhspot.com/mhspot/sippyskype.htm as usual.

If you find a Skype client that supports multiple active calls, please post the version and platform.

SippySkype 20080628 Sip To Skype Bridge Update

Latest changes:
  • Added new parameter autoShutdownMinutes - shutdown after time has expired and no call in progress - 5 minute increments
  • Added support for DTMF over INFO messages
  • Added new parameter dtmfinfotype - set to application/dtmf or application/dtmf-relay to activate DTMF over INFO msgs
  • Added enableSkypeDtmfDetector to allow experiments with decoding of skype analog dtmf
  • Fixed blank pin not working

Download at http://www.mhspot.com/mhspot/sippyskype.htm as usual.

SippySkype 20080616 Skype to SIP Gateway Update

Latest changes:
  • Moved OPTIONS handler code to a earlier location to guarantee message gets intercepted.
  • Fixed some more java 1.5 compatibility issues.
  • Added 486 response to invite if no INVITE listener available.
  • Added new parameter skypeimmessage to set your own IM message.
  • Added init contact info if none specified.

Download at http://www.mhspot.com/mhspot/sippyskype.htm as usual.

SippySkype 20080614 Skype/SIP Gateway Update


Latest changes:
  • Added new parameter dtmf2833payloadtype - default is still 101
  • Change RTP delay timing to adjust for cpu loading - affects playing of files and authentication only
  • Changed blank RTP packets to contain a valid value.
  • Fix some issues with SIP dtmf sending code.
  • Added cnonce/nc support if qop authentication specified
  • Reduce java requirements to java 1.5
  • Fix intermittant loss of INVITE handler - It would stop answering SIP calls
  • Added refuse of multiple incoming skype calls since it isn't supported.

Download at http://www.mhspot.com/mhspot/sippyskype.htm as usual.

SippySkype 20080606 Update

  • Added SIP OPTIONS message support so Asterisk can determine online status
  • Reduced likelihood of getting a duplicate branch
  • Fix REGISTER www authenication not creating new branch
  • Fix INVITE www/proxy authenication not creating new branch
  • Added new parameter sendSkypeIM to send an incoming call message when calling a skype user

Download at http://www.mhspot.com/mhspot/sippyskype.htm as usual.

SippySkype Update 20080601

Lastest Changes:
  • Removed setting of SkypeUserid in SIP UserAgent. May help with Asterisk authorization issues.

Download at http://www.mhspot.com/mhspot/sippyskype.htm as usual.

SippySkype 20080528 update

Lastest Changes:
  • Fixed Linux/OSX startup script
  • Added experiental handleEarlyMedia flag to send Skype status audio during SkypeOut calls

Download at http://www.mhspot.com/mhspot/sippyskype.htm as usual.

SippySkype Update 20080427

Lastest Changes:
  • added ack for 401/407 response to invite to help with Asterisk 491 error

Otherwise same as 20080406

Download at http://www.mhspot.com/mhspot/sippyskype.htm as usual.

SippySkype Update 20080406

Lastest Changes:
  • Change to mjua to reset local session when handling calls - it was using last incoming SIP call origins as it's own for outgoing SIP calls.
  • Changed origin to set a session id.
  • Removed user name prefix from origin owner
  • Fixed real registration problem. When provider sends multiple expires tags, mjsip was grabbing the wrong one. Changed to select the highest which should be the most recent registration.

Download at http://www.mhspot.com/mhspot/sippyskype.htm as usual.