Asterisk 11.3.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 11.3.0. This release is available for immediate download at

The release of Asterisk 11.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following are the issues resolved in this release:

  • [ASTERISK-15456] – chan_misdn does not set INVALID_EXTEN
  • [ASTERISK-16357] – chan_mobile unable to connect to cellphone
  • [ASTERISK-16610] – problem to reload the module skinny when active calls.
  • [ASTERISK-16822] – Channel Variable SMSSRC not set properly
  • [ASTERISK-16854] – roundf causing asterisk to fail to compile
  • [ASTERISK-18697] – [minivm] Crash in MinivmNotify
  • [ASTERISK-18975] – Manager Redirect action on bridged channel pair causes intermittent hangup on second channel
  • [ASTERISK-19153] – – Sms sender is not parsed correctly in incoming sms
  • [ASTERISK-19948] – Asterisk 1.8 manager redirect command fails when redirecting multiple channels currently bridged together via dial command.
  • [ASTERISK-20407] – Asterisk compilation doesn’t set rpath when –prefix is something other that /usr
  • [ASTERISK-20440] – No ringback towards SLAstation on outbound trunk call.
  • [ASTERISK-20462] – Trunk not hungup if SLA Station hangs up before answer
  • [ASTERISK-20606] – Wrong confbridge behavior when participants enter simultaneously
  • [ASTERISK-20653] – Asterisk allows Session-Expires below 90 in a 200 OK
  • [ASTERISK-20708] – Deadlock in chan_sip on transfer when trying to update redirecting information
  • [ASTERISK-20716] – “s” extension in comebackcontext not honored
  • [ASTERISK-20717] – Voicemail access “SQL Get Data error! coltitle=msg_id”
  • [ASTERISK-20743] – Queue Log – All Calls End With COMPLETECALLER When h Extension Is Present
  • [ASTERISK-20772] – Loop bug in ast_rtp_lookup_mime_multiple2() [main/rtp_engine.c]
  • [ASTERISK-20789] – Make skinny debug tab completion helpful
  • [ASTERISK-20790] – skinny does not respect globally set vmexten
  • [ASTERISK-20801] – Non-SIP queue members get no calls when ringinuse=no.
  • [ASTERISK-20805] – SIP Notify message has incorrect IP address in FROM field
  • [ASTERISK-20837] – build_route fails to parse Record-Route headers longer than 255 characters
  • [ASTERISK-20849] – SDP crypto attribute is not well formed in the SDP ANSWER
  • [ASTERISK-20852] – asterisk/strings.h: struct ast_str used before its declaration
  • [ASTERISK-20897] – case sensitive match against T.38 params causes T38MaxBitRate to be negotiated at 2400 baud instead of 14400
  • [ASTERISK-20898] – sound_only_one parameter will be ignored in confbridge.conf
  • [ASTERISK-20906] – DTMF in SIP not working after HOLD / UNHOLD
  • [ASTERISK-20908] – Asterisk presents media desc for video in SDP, missing terminating CRLF
  • [ASTERISK-20914] – Segfault when iLBC voice frame is interpolated in a jitter buffer due to codec_ilbc’s improper manipulation of datalen
  • [ASTERISK-20916] – GoogleVoice calls don’t connect, but continue ringing despite call having been answered
  • [ASTERISK-20938] – ConfBridge list from CLI and Manager no longer include waiting members
  • [ASTERISK-20945] – “Unable to connect to remote asterisk” message on service asterisk start, even though service is running
  • [ASTERISK-20947] – astcanary exits immediately because of wrong pid argument
  • [ASTERISK-20964] – Device call logging has issues.
  • [ASTERISK-20980] – ./configure fails with ptlib 2.10.9
  • [ASTERISK-21006] – unsupported host os “linux-gnueabihf”
  • [ASTERISK-21012] – Memory Leak on res_calendar (icalendar)
  • [ASTERISK-21080] – Redial button does not work properly
  • [ASTERISK-21298] – Confbridge recording fails – deadlock

For a full list of changes in this release, please see the ChangeLog:

Thank you for your continued support of Asterisk!

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