Asterisk 13.34.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.34.0.

This release is available for immediate download at

https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.34.0 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:

———————————–

res_pjsip_logger writing too big packets
(Reported by nappsoft)
Wrong return value check for fwrite when writing to pcap file
(Reported by nappsoft)
res_pjsip: Crash when escaping during URI printing
(Reported by nappsoft)
x-ast-orig-host not filtered out from request URI and To header
(Reported by nappsoft)
bridge_softmix: Conference bridge not passing silent rtp packets
(Reported by Jonathan Hunter)
RTP ICE leaks the memory
(Reported by sungtae kim)
SIGSEGV when pjsip show history encounters IPV6 address
(Reported by Roger James)
tcptls: Fix notice when TLS is enabled but not configured.
(Reported by Alexander Traud)
app_osplookup.c: Avoid a format truncation.
(Reported by Alexander Traud)
Non async-signal-safe syscalls used after fork before exec
(Reported by nappsoft)
app_queue: leaking stasis subscription when Redirecting call
(Reported by lvl)
app_queue: Ghost channels in “core show channels” output
(Reported by Etienne Lessard)
Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling
(Reported by Shlomi Gutman)
pjsip: Increase maximum candidate count
(Reported by Joshua C. Colp)
Unprotected access to nochecksums variable, causes build failures
(Reported by Guido Falsi)

Improvements made in this release:

———————————–

res_pjsip_logger: Add tons’o’functionality
(Reported by Joshua C. Colp)
pjproject has race conditions in it’s build system
(Reported by Guido Falsi)
third-party/pjproject/configure.m4 contains bashisms
(Reported by Guido Falsi)
chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio
(Reported by Peter Turczak)

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

https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.34.0

Thank you for your continued support of Asterisk!

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