Posts Tagged CCSS

libpri 1.4.12 Now Available

Posted by admin in libpri on July 6, 2011

libpri

libpri

The Asterisk Development Team announces the release of libpri version 1.4.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/

The following are some of the issues resolved in this release:

  • Add call transfer exchange of subaddresses support and fix PTMP call transfer signaling.
  • Invalid PTMP redirecting signaling as TE towards NT.
  • Add Q931_IE_TIME_DATE to CONNECT message when in network mode.
    (issue #18047 (JIRA PRI-114). Reported by: wuwu. Patched by rmudgett)
  • Swap of master/slave in pri_enslave() incorrect.
    (issue #18769 (JIRA PRI-120). Reported by: jcollie. Patched by jcollie)
  • Fix I-frame retransmission quirks.
  • Crash if NFAS swaps D channels on a call with an active timer.
  • DMS-100 not receiving caller name anymore.
    (issue #18822 (JIRA PRI-121). Reported by: cmorford. Patched by rmudgett)
  • B channel lost by incoming call in BRI NT PTMP mode.
  • Implement the mandatory T312 timer for NT PTMP broadcast SETUP calls.

This release contains several new features, among them:

  1. ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
  2. ETSI Advice Of Charge (AOC) support
  3. ETSI Explicit Call Transfer (ECT) support
  4. ETSI Call Waiting support for ISDN phones
  5. ETSI Malicious Call ID support
  6. Add Display IE text handling options.

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

http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1….

Thank you for your continued support of Asterisk!

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Asterisk 1.8.4-rc2 Now Available

Posted by admin in asterisk, Release Candidates, sip on February 28, 2011

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the second release candidate of Asterisk 1.8.4. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.4-rc2 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

  • Resolution of several DTMF based attended transfer issues.
    (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett)
    NOTE: Be sure to read the ChangeLog for more information about these changes.
  • Resolve deadlocks related to device states in chan_sip
    (Closes issue #18310. Reported, patched by one47. Patched by jpeeler)
  • Resolve an issue with the Asterisk manager interface leaking memory when disabled.
    (Reported internally by kmorgan. Patched by russellb)
  • Support greetingsfolder as documented in voicemail.conf.sample.
    (Closes issue #17870. Reported by edhorton. Patched by seanbright)
  • Fix channel redirect out of MeetMe() and other issues with channel softhangup
    (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb)
  • Fix voicemail sequencing for file based storage.
    (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler)
  • Set hangup cause in local_hangup so the proper return code of 486 instead of 503 when using Local channels when the far sides returns a busy. Also affects CCSS in Asterisk 1.8+.
    (Patched by twilson)
  • Fix issues with verbose messages not being output to the console.
    (Closes issue #18580. Reported by pabelanger. Patched by qwell)

Asterisk 1.8.4-rc1 was not released due to a blocking issue found prior to release. An additional fix was merged into Asterisk 1.8.4-rc2:

  • Fix Deadlock with attended transfer of SIP call
    (Closes issue #18837. Reported, patched by alecdavis. Tested by alecdavid, Irontec, ZX81, cmaj)

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

Thank you for your continued support of Asterisk!

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Libpri 1.4.12-Beta1 Now Available

Posted by admin in libpri on July 27, 2010

The Asterisk Development Team has announced the release of libpri version 1.4.12-beta1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/

This beta release contains some fixes and several new features, among them:

  1. ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
  2. ETSI Advice Of Charge (AOC) support
  3. ETSI Explicit Call Transfer (ECT) support
  4. ETSI Call Waiting support for ISDN phones
  5. ETSI Malicious Call ID support

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

http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.12-beta1

Thank you for your continued support of Asterisk!

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