Asterisk 13.24.0-rc1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.24.0-rc1.
This release is available for immediate download at

The release of Asterisk 13.24.0-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:

Security bugs fixed in this release:


res_http_websocket: Crash when reading HTTP Upgrade requests
(Reported by Sean Bright)


New Features made in this release:


add flag to allow CALLERID(num) to be placed in Contact header in chan_pjsip
(Reported by Torrey Searle)


Bugs fixed in this release:


app_queue: Revert broken queue channel reference patch
(Reported by lvl)
app_voicemail: MWI fails with mailboxes=##@device instead of mailboxes=##@default
(Reported by Ronald Raikes)
Asterisk crashes when the res_pjsip_* modules unload
(Reported by sungtae kim)
SIGABRT caused by stack corruption in hashkeys_read when no matching keys present
(Reported by Michael Walton)
repeated segmentation faults
(Reported by Eyal Hasson)
stasis: Filter messages at publishing to reduce work done
(Reported by Joshua C. Colp)
Incorrect Behavior for rewrite_contact when Re-Invite omits routset
(Reported by Torrey Searle)
Some conditions prevent running of el_end, break the terminal.
(Reported by Corey Farrell)
need to reset DTMF last sequence number and timestamp on voice packet with marker bit
(Reported by Alexei Gradinari)
rtp: Incorrect Packetization
(Reported by Robert Cripps)
pbx_config: Only the first [globals] section is processed.
(Reported by Corey Farrell)
Formatting error in documentation
(Reported by Scott Griepentrog)
chan_sip: Asterisk 12+ chan_sip doesn’t report AST_CEL_PICKUP in handle_invite_replaces
(Reported by Luit van Drongelen)
res_pjsip_notify: improve realtime performance on CLI completion on the endpoint
(Reported by Alexei Gradinari)
Caller ID cannot be changed on Attended Transfer before dialing out
(Reported by Alexei Gradinari)
function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload
(Reported by Emmanuel BUU)
bridging: Asterisk crashes when receiving an empty realtime text frame
(Reported by Emmanuel BUU)
app_queue: QueueMemberStatus Event flooding AMI
(Reported by Andrej)
res_pjsip: improve realtime performance on CLI ‘pjsip show contacts’
(Reported by Alexei Gradinari)
stasis: Playing MOH to bridge with ARI does not work
(Reported by Cameron)
app_queue: Queue member considered inuse after immediately hanging up during dialing.
(Reported by Cao Minh Hiep)
testsuite: Sniffer assumes pjmedia will use ports below 10000
(Reported by Joshua C. Colp)
res_odbc: missing SQL error diagnostic
(Reported by Alexei Gradinari)
res_pjsip_mwi: Memory leak on reload
(Reported by Sergej Kasumovic)
PJSIP: Update bundled PJPROJECT to version 2.8
(Reported by Joshua C. Colp)
chan_sip: SipNotify via AMI behaves differently to CLI
(Reported by Peter Katzmann)
res_pjproject build failure
(Reported by Jaco Kroon)
res_musiconhold : music on hold will not start if previous hold just reached end of file
(Reported by Frederic LE FOLL)
Realtime queuemembers are not updated during retry phase
(Reported by lvl)
alembic: PJSIP “mwi_subscribe_replaces_unsolicited” field is integer not boolean
(Reported by Joshua C. Colp)
res_pjsip_transport_websocket: Properly set ‘received’ for IPv6
(Reported by Sean Bright)


Improvements made in this release:


New function PJSIP_PARSE_URI to parse an URI and return a specified part of the URI
(Reported by Alexei Gradinari)
Allow the sip_to_pjsip script to be used in a pipe
(Reported by Pascal Cadotte Michaud)
Remove stale nonoptreq references
(Reported by Walter Doekes)


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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