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Releases: tayler6000/pyVoIP

pyVoIP v1.6.5-rc.1

07 Jun 14:14
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pyVoIP v1.6.5-rc.1 Pre-release
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What's Changed

Full Changelog: v1.6.4...v1.6.5-rc.1

pyVoIP v1.6.4

21 Jan 19:18
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Fixed registration issues with some PBXs such as Fritzbox. Patch credited to @hartwigt who cited Philipp Fischer (@philipp-fischer) and Nicolas Fleiter (@epicbananana) here.

Full Changelog: v1.6.3...v1.6.4

pyVoIP v1.6.3

16 Jan 18:17
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Full Changelog: v1.6.2...v1.6.3

pyVoIP v1.6.2

26 Sep 15:44
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What's Changed

  • [CHANGE] new parsing of WWW-Authenticate header by @yTI0r in #67
  • [FIX] Fix regex for comma-separated values by @dermodmaster in #68

New Contributors

Full Changelog: v1.6.0...v1.6.2

pyVoIP v1.6.0

19 Sep 19:27
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What's Changed

New Contributors

Full Changelog: v1.5.6...v1.6.0

pyVoIP v1.5.6

12 Sep 18:33
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[ADD] Added timeout checking to registration and deregistration.
[FIX] Fixed minor bugs.

pyVoIP v1.5.5

16 Apr 13:56
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[FIX] Fixed #27
[FIX] Fixed #38
[FIX] Fixed improper error raise in SIPClient.start()
[ADD] Added NoPortsAvailableError in VoIP
[ADD] Added VoIPPhone.request_port(blocking=True).
If blocking is set to false, NoPortsAvailableError may be raised,
otherwise will wait.
[ADD] Added VoIPPhone.release_ports(call: Optional[VoIPCall] = None).
If call is provided, release ports assinged to call,
Otherwise release all ports not in use.
[ADD] Added VoIPPhone._cleanup_dead_calls(). It handles dead threads.
[CHANGE] Changed VoIPCall to not take portRange. It is now pulled from VoIPPhone
[CHANGE] Changed all instances of port assignment to VoIPPhone.request_port()
[CHANGE] Changed VoIPPhone.start() to except BaseException instead of Exception
[CHANGE] Changed docs to reflect changes.

pyVoIP v1.5.4

02 Mar 06:42
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Changing to 1.5.4 due to PyPI conflict

pyVoIP v1.5.3

16 Feb 00:08
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Fixes in v1.5.3:
[FIXED] Fixed #18
[FIXED] Fixed #19

pyVoIP v1.5.2

03 Feb 02:27
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Fixes in v1.5.2:
[FIXED] Fixed issue #10.
[FIXED] Fixed documentation typos.

Fixes in v1.5.1:
Fixed issue #2 pointed out by rushant001.

New in v1.5.0:
Fixed bug where pyVoIP would accept all codecs proposed by the server even if not compatible. Will now only accept PCMU, PCMA, and telephone-event.
Added handling of Native Bridging tested with Asterisk 16 SIP re-invite (External RTP bridge), this seems to still have issues with Asterisk 18, but unsure if it's my hardphone.
Changed the audio read function in RTP to return b'\x80'*length instead of bytes(length), doing so stops the popping on the client-side when no audio is being written.
Fixed issue with ending phone calls originated by user.
Added handling of 404 Not Found and 503 Service Unavailable errors.
Added compatibility with Asterisk PJSIP.
Fixed bug with multithreaded calling.