Asterisk 1.8.32.0-rc1 Now Available

The Asterisk Development Team has announced the first release candidate of Asterisk 1.8.32.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.32.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release candidate:

Bug

• [ASTERISK-13797] – relax badshell tilde test
• [ASTERISK-15879] – Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
• [ASTERISK-18923] – res_fax_spandsp usage counter is wrong
• [ASTERISK-20784] – Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
• [ASTERISK-21721] – SIP Failed to parse multiple Supported: headers
• [ASTERISK-22791] – asterisk sends Re-INVITE after receiving a BYE
• [ASTERISK-22945] – Memory leaks in chan_sip.c with realtime peers
• [ASTERISK-23768] – Asterisk man page contains a (new) unquoted minus sign
• [ASTERISK-23846] – Unistim multilines. Loss of voice after second call drops (on a second line).
• [ASTERISK-24011] – safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM
• [ASTERISK-24063] – Asterisk does not respect outbound proxy when sending qualify requests
• [ASTERISK-24190] – IMAP voicemail causes segfault
• [ASTERISK-24325] – res_calendar_ews: cannot be used with neon 0.30
• [ASTERISK-24335] – [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls
• [ASTERISK-24348] – Built-in editline tab complete segfault with MALLOC_DEBUG
• [ASTERISK-24357] – [fax] Out of bounds error in update_modem_bits
• [ASTERISK-24390] – astobj2: REF_DEBUG reports false leaks with ao2_callback with OBJ_MULTIPLE
• [ASTERISK-24393] – rtptimeout=0 doesn’t disable rtptimeout
• [ASTERISK-24406] – Some caller ID strings are parsed differently since 11.13.0
• [ASTERISK-24425] – jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
• [ASTERISK-24432] – Install refcounter.py when REF_DEBUG is enabled
• [ASTERISK-24436] – Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
• [ASTERISK-24476] – main/app.c / app_voicemail: ast_writestream leaks

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.32.0-rc1

Thank you for your continued support of Asterisk!