Asterisk 15.6.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
This release is available for immediate download at

The release of Asterisk 15.6.0 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:

Bugs fixed in this release:


When T.140 realtime text is negociated, a lot of debug traces are generated
(Reported by Emmanuel BUU)
PBX calls via chan_sip TCP trunk now get authentification error
(Reported by Ian Gilmour)
chan_sip: get_refer_info() attempted unlock mutex ‘peer’ without owning it!
(Reported by Alec Davis)
res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE
(Reported by Joshua Elson)
rtcp-mux is put in SDP answer regardless of offer
(Reported by Torrey Searle)
No joint capabilities with video and audio-only streams
(Reported by Benjamin Keith Ford)
(Reported by Valentin Safonov)
pjproject_bundled: Fix for Solaris builds. Do not undef s_addr.
(Reported by Alexander Traud)
Wrong SRTP use status report
(Reported by Salah Ahmed)
res_pjsip_registrar: Improve performance of inbound handling
(Reported by Joshua Colp)
pjsip: Race condition in 183 re transmission can result in a deadlock
(Reported by Torrey Searle)
make menuselect fails due to undefined symbols (initscr32, w32addch) in menuselect_curses.o
(Reported by Majdi Bsoul)
[regression] menuselect compilation failure on Solaris 10
(Reported by Samuel Owens)
menuselect compilation failure on Solaris 10 / gcc 3.4.3
(Reported by rleasure)
menuselect compilation failure on Solaris 10/gcc-4.1.1
(Reported by Bob Atkins)
BuildSystem: Enable Jansson in Solaris 11.
(Reported by Alexander Traud)
res_pjsip_endpoint_identifier_ip only matches against “generic string” headers
(Reported by George Joseph)
res_rtp_asterisk: Requires OpenSSL in Developer Mode.
(Reported by Alexander Traud)
Frack errors in stasis.c and memory leakage
(Reported by Siruja Maharjan)
res_pjsip: Change default transport keepalive to preserve behavior
(Reported by Joshua Colp)
systemd: asterisk.service
(Reported by seanchann.zhou)
pjproject_bundled: Repair ./configure –with-ssl=PATH.
(Reported by Alexander Traud)
BASIC-RETRANS: Implement receive
(Reported by Benjamin Keith Ford)
res_sorcery_config: Allow object name based matching
(Reported by Joshua Colp)
stasis: Improve message type “Use of before init/after destruction” error
(Reported by Joshua Colp)
srtp: rejecting short sdes lifetimes incompatible with obihai ATAs
(Reported by Nick French)
res_pjsip: Spurious ERROR logging when printing headers in sip_msg
(Reported by Nick French)
pjsip modules always get -O2 even when DONT_OPTIMIZE is set
(Reported by George Joseph)
PJSIP proposes ICE candidates on answer even if not in offer
(Reported by Torrey Searle)
pjproject_bundled: Disable TCP/TLS keep-alives.
(Reported by Alexander Traud)
Compile fails with `IPTOS_MINCOST’ undeclared.
(Reported by Alexander Traud)
res_pjsip_session: sdp group:BUNDLE attribute truncated
(Reported by Kevin Harwell)
res_pjsip_pubsub: segfault in function publish_expire
(Reported by Alexei Gradinari)
res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason headers
(Reported by Ross Beer)
res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of declining stream
(Reported by Thiago Coutinho)
res_pjsip_t38: ATA fails with hangupcause 58(Bearer capability not available)
(Reported by Jared Hull)
res_pjsip_t38: Slow T.38 re-invite rejection if remote leg has T.38 disabled
(Reported by Torrey Searle)
res_pjsip: Lock inversion in transport management
(Reported by Ross Beer)
bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
(Reported by Alexander Traud)


Improvements made in this release:


PJSIP: Missing “party=calling”/”party=called” in Remote-Party-ID
(Reported by Eric Dantie)
pjproject_bundled: Find shared libraries in root –with-ssl=PATH.
(Reported by Alexander Traud)
pjsip_wizard example gives wrong info about unsupported SRV records
(Reported by Jonathan Harris)
res_rtp_asterisk: T.140 packets containing backspace or end of line are merged with regular text and it causes some UA to break
(Reported by Emmanuel BUU)


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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