The Asterisk Development Team would like to announce the first
release candidate of Asterisk 13.29.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.29.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release candidate:
Bugs fixed in this release:
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ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) |
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chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) |
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codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) |
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translate: Crash when frame does not have a “src” field set (Reported by Gregory Massel) |
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chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) |
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pjsip mwi: n+1 sip notify’s sent on re-register (Reported by Chris Savinovich) |
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PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters (Reported by Dan Cropp) |
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app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream (Reported by Alexei Gradinari) |
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compile menuselect on gentoo (Reported by Kilburn) |
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Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV (Reported by Jonas Swiatek) |
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cel / cdr: Event times may be incorrect (Reported by Joshua C. Colp) |
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packet lost on UDPTL wrap around (Reported by Torrey Searle) |
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json integer overflow in ssrc and timestamp (Reported by Salah Ahmed) |
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res_pjsip: pjsip show contacts prints double entries (Reported by Ian Jones) |
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Crash when not specifying “dbfile” in res_config_sqlite3.conf (Reported by Dennis) |
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Crash performing “core reload” with modified res_config_sqlite3.conf (Reported by Dennis) |
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AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) (Reported by Walter Doekes) |
New Features made in this release:
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Unregister a realtime moh class (Reported by Byron Clark) |
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Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) |
For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.29.0-rc1
Thank you for your continued support of Asterisk!
