[asterisk: 8/9] Update to 1.8.2.2 and enable SRTP
Jeffrey C. Ollie
jcollie at fedoraproject.org
Mon Jan 24 20:37:55 UTC 2011
commit 64989e248a723f77b28d55c14902efc55e263ebb
Author: Jeffrey C. Ollie <jeff at ocjtech.us>
Date: Mon Jan 24 14:04:56 2011 -0600
Update to 1.8.2.2 and enable SRTP
.gitignore | 6 +
...igure.ac-to-look-for-pkg-config-gmime-2.0.patch | 35 ---
0006-Fix-up-some-paths.patch | 95 -------
asterisk.spec | 264 +++++++++++++++++---
menuselect.makeopts | 4 +-
sources | 4 +-
6 files changed, 233 insertions(+), 175 deletions(-)
---
diff --git a/.gitignore b/.gitignore
index 6dfc296..835f42b 100644
--- a/.gitignore
+++ b/.gitignore
@@ -16,3 +16,9 @@ asterisk-1.8.0-beta3.tar.gz.asc
/asterisk-1.8.0-rc5.tar.gz.asc
/asterisk-1.8.0.tar.gz
/asterisk-1.8.0.tar.gz.asc
+/asterisk-1.8.1-rc1.tar.gz
+/asterisk-1.8.1-rc1.tar.gz.asc
+/asterisk-1.8.1.tar.gz
+/asterisk-1.8.1.tar.gz.asc
+/asterisk-1.8.2.2.tar.gz
+/asterisk-1.8.2.2.tar.gz.asc
diff --git a/0006-Fix-up-some-paths.patch b/0006-Fix-up-some-paths.patch
index da802f7..58703d6 100644
--- a/0006-Fix-up-some-paths.patch
+++ b/0006-Fix-up-some-paths.patch
@@ -180,101 +180,6 @@ index 063feca..9ace9f2 100644
.RS
.RE
-diff --git a/doc/osp.txt b/doc/osp.txt
-index 9d059f0..f07e9f0 100644
---- a/doc/osp.txt
-+++ b/doc/osp.txt
-@@ -136,7 +136,7 @@ make clean; make linux
- Compilation is successful if there are no errors in the compiler output. The enroll program is now located in the OSP Toolkit/bin directory (example: /usr/src/ TK-3_3_6-20060303/bin).
-
- 2.2 Obtain Crypto Files
--The OSP module in Asterisk requires three crypto files containing a local certificate (localcert.pem), private key (pkey.pem), and CA certificate (cacert_0.pem). Asterisk will try to load the files from the Asterisk public/private key directory - /var/lib/asterisk/keys. If the files are not present, the OSP module will not start and the Asterisk will not support the OSP protocol. Use the enroll.sh script from the toolkit distribution to enroll Asterisk with an OSP server and obtain the crypto files. Documentation explaining how to use the enroll.sh script (Device Enrollment) to enroll with an OSP server is available at http://www.transnexus.com/OSP%20Toolkit/OSP%20Toolkit%20Documents/Device_Enrollment.pdf. Copy the files generated by the enrollment process to the Asterisk /var/lib/asterisk/keys directory.
-+The OSP module in Asterisk requires three crypto files containing a local certificate (localcert.pem), private key (pkey.pem), and CA certificate (cacert_0.pem). Asterisk will try to load the files from the Asterisk public/private key directory - /usr/share/asterisk/keys. If the files are not present, the OSP module will not start and the Asterisk will not support the OSP protocol. Use the enroll.sh script from the toolkit distribution to enroll Asterisk with an OSP server and obtain the crypto files. Documentation explaining how to use the enroll.sh script (Device Enrollment) to enroll with an OSP server is available at http://www.transnexus.com/OSP%20Toolkit/OSP%20Toolkit%20Documents/Device_Enrollment.pdf. Copy the files generated by the enrollment process to the Asterisk /usr/share/asterisk/keys directory.
-
- Note: The osptestserver.transnexus.com is configured only for sending and receiving non-SSL messages, and issuing signed tokens. If you need help, post a message on the OSP mailing list at https://lists.sourceforge.net/lists/listinfo/osp-toolkit-client..
-
-@@ -183,7 +183,7 @@ depth=0 /CN=osptestserver.transnexus.com/O=OSPServer
- verify return:1
- The certificate request was successful.
- Error Code returned from localcert command : 0
--The files generated should be copied to the /var/lib/asterisk/keys directory.
-+The files generated should be copied to the /usr/share/asterisk/keys directory.
- Note: The script enroll.sh requires AT&T korn shell (ksh) or any of its compatible variants. The /usr/src/TK-3_3_6-20060303/bin directory should be in the PATH variable. Otherwise, enroll.sh cannot find the enroll file.
-
- 3 Asterisk
-@@ -247,9 +247,9 @@ servicepoint=http://OSP server IP:5045/osp
- source=[host IP]
- ;
- ; Define path and file name of crypto files.
--; The default path for crypto file is /var/lib/asterisk/keys. If no
-+; The default path for crypto file is /usr/share/asterisk/keys. If no
- ; path is defined, crypto files should be in
--; /var/lib/asterisk/keys directory.
-+; /usr/share/asterisk/keys directory.
- ;
- ; Specify the private key file name.
- ; If this parameter is unspecified or not present, the default name
-diff --git a/doc/tex/asterisk-conf.tex b/doc/tex/asterisk-conf.tex
-index 4b08023..3e35f92 100644
---- a/doc/tex/asterisk-conf.tex
-+++ b/doc/tex/asterisk-conf.tex
-@@ -18,10 +18,10 @@ astetcdir => /etc/asterisk
- astmoddir => /usr/lib/asterisk/modules
-
- ; Where additional 'library' elements (scripts, etc.) are located
--astvarlibdir => /var/lib/asterisk
-+astvarlibdir => /usr/share/asterisk
-
- ; Where AGI scripts/programs are located
--astagidir => /var/lib/asterisk/agi-bin
-+astagidir => /usr/share/asterisk/agi-bin
-
- ; Where spool directories are located
- ; Voicemail, monitor, dictation and other apps will create files here
-diff --git a/doc/tex/phoneprov.tex b/doc/tex/phoneprov.tex
-index 790c1d7..b30b5c5 100644
---- a/doc/tex/phoneprov.tex
-+++ b/doc/tex/phoneprov.tex
-@@ -58,7 +58,7 @@ files, respectively. A sample profile:
- [polycom]
- staticdir => configs/
- mime_type => text/xml
--setvar => CUSTOM_CONFIG=/var/lib/asterisk/phoneprov/configs/custom.cfg
-+setvar => CUSTOM_CONFIG=/usr/share/asterisk/phoneprov/configs/custom.cfg
- static_file => bootrom.ld,application/octet-stream
- static_file => bootrom.ver,plain/text
- static_file => sip.ld,application/octet-stream
-diff --git a/doc/tex/privacy.tex b/doc/tex/privacy.tex
-index a4ae7b9..bad6815 100644
---- a/doc/tex/privacy.tex
-+++ b/doc/tex/privacy.tex
-@@ -210,7 +210,7 @@ helpful.
-
- When there is no CallerID, P and p options will always record an intro
- for the incoming caller. This intro will be stored temporarily in the
--\path{/var/lib/asterisk/sounds/priv-callerintros} dir, under the name
-+\path{/usr/share/asterisk/sounds/priv-callerintros} dir, under the name
- NOCALLERID\_$<$extension$>$ $<$channelname$>$ and will be erased after the
- callee decides what to do with the call.
-
-@@ -245,7 +245,7 @@ introductions are stored and re-used for the convenience of the CALLER.
- \subsubsection{Introductions}
- Unless instructed to not save introductions (see the 'n' option above),
- the screening modes will save the recordings of the caller's names in
--the directory \path{/var/lib/asterisk/sounds/priv-callerintros}, if they have
-+the directory \path{/usr/share/asterisk/sounds/priv-callerintros}, if they have
- a CallerID. Just the 10-digit callerid numbers are used as filenames,
- with a ".gsm" at the end.
-
-@@ -260,7 +260,7 @@ loudspeakers, and perhaps other nifty things. For instance:
-
- \begin{astlisting}
- \begin{verbatim}
--exten => s,6,Set(PATH=/var/lib/asterisk/sounds/priv-callerintros)
-+exten => s,6,Set(PATH=/usr/share/asterisk/sounds/priv-callerintros)
- exten => s,7,System(/usr/bin/play ${PATH}/${CALLERID(num)}.gsm&,0)
- \end{verbatim}
- \end{astlisting}
diff --git a/pbx/ael/ael-test/ael-test3/extensions.ael b/pbx/ael/ael-test/ael-test3/extensions.ael
index ff1f6ae..53fb918 100755
--- a/pbx/ael/ael-test/ael-test3/extensions.ael
diff --git a/asterisk.spec b/asterisk.spec
index c247b3f..b61a855 100644
--- a/asterisk.spec
+++ b/asterisk.spec
@@ -1,9 +1,9 @@
-#global _rc 5
+#global _rc 1
#global _beta 5
Summary: The Open Source PBX
Name: asterisk
-Version: 1.8.0
-Release: 3%{?_rc:.rc%{_rc}}%{?_beta:.beta%{_beta}}%{?dist}
+Version: 1.8.2.2
+Release: 2%{?_rc:.rc%{_rc}}%{?_beta:.beta%{_beta}}%{?dist}
License: GPLv2
Group: Applications/Internet
URL: http://www.asterisk.org/
@@ -19,11 +19,8 @@ Patch2: 0002-Modify-modules.conf-so-that-different-voicemail-modu.patch
# Submitted upstream: https://issues.asterisk.org/view.php?id=16858
Patch3: 0003-Allow-linking-building-against-an-external-libedit.patch
Patch4: 0004-Use-the-library-function-for-loading-command-history.patch
-# Submitted upstream: https://issues.asterisk.org/view.php?id=16155
-Patch5: 0005-Change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch
Patch6: 0006-Fix-up-some-paths.patch
Patch7: 0007-Add-LDAP-schema-that-is-compatible-with-Fedora-Direc.patch
-Patch8: 0008-Tell-laxtex2html-to-copy-icons-when-building-documen.patch
BuildRoot: %{_tmppath}/%{name}-%{version}-root-%(%{__id_u} -n)
@@ -39,6 +36,7 @@ BuildRequires: libtermcap-devel
BuildRequires: ncurses-devel
BuildRequires: libcap-devel
BuildRequires: gtk2-devel
+BuildRequires: libsrtp-devel
# for res_http_post
%if 0%{?fedora} > 0
@@ -431,10 +429,8 @@ local filesystem.
%patch2 -p1
%patch3 -p1
%patch4 -p1
-%patch5 -p1
%patch6 -p1
%patch7 -p1
-%patch8 -p1
cp %{S:3} menuselect.makedeps
cp %{S:4} menuselect.makeopts
@@ -477,9 +473,9 @@ pushd menuselect
popd
%if 0%{?fedora} > 0
-%configure --with-imap=system --with-gsm=/usr --with-libedit=yes
+%configure --with-imap=system --with-gsm=/usr --with-libedit=yes --with-srtp
%else
-%configure --with-gsm=/usr --with-libedit=yes
+%configure --with-gsm=/usr --with-libedit=yes --with-srtp
%endif
ASTCFLAGS="%{optflags}" make DEBUG= OPTIMIZE= ASTVARRUNDIR=%{_localstatedir}/run/asterisk ASTDATADIR=%{_datadir}/asterisk ASTVARLIBDIR=%{_datadir}/asterisk ASTDBDIR=%{_localstatedir}/spool/asterisk NOISY_BUILD=1
@@ -525,8 +521,8 @@ install -D -p -m 0755 contrib/init.d/rc.redhat.asterisk %{buildroot}%{_initrddir
install -D -p -m 0644 contrib/sysconfig/asterisk %{buildroot}%{_sysconfdir}/sysconfig/asterisk
install -D -p -m 0644 contrib/scripts/99asterisk.ldif %{buildroot}%{_sysconfdir}/dirsrv/schema/99asterisk.ldif
install -D -p -m 0644 %{S:2} %{buildroot}%{_sysconfdir}/logrotate.d/asterisk
-install -D -p -m 0644 doc/asterisk-mib.txt %{buildroot}%{_datadir}/snmp/mibs/ASTERISK-MIB.txt
-install -D -p -m 0644 doc/digium-mib.txt %{buildroot}%{_datadir}/snmp/mibs/DIGIUM-MIB.txt
+#install -D -p -m 0644 doc/asterisk-mib.txt %{buildroot}%{_datadir}/snmp/mibs/ASTERISK-MIB.txt
+#install -D -p -m 0644 doc/digium-mib.txt %{buildroot}%{_datadir}/snmp/mibs/DIGIUM-MIB.txt
rm %{buildroot}%{_libdir}/asterisk/modules/app_directory.so
rm %{buildroot}%{_libdir}/asterisk/modules/app_voicemail.so
@@ -593,21 +589,21 @@ fi
%doc README* *.txt ChangeLog BUGS CREDITS configs
%doc doc/asterisk.sgml
-%doc doc/backtrace.txt
-%doc doc/callfiles.txt
-%doc doc/externalivr.txt
-%doc doc/macroexclusive.txt
-%doc doc/manager_1_1.txt
-%doc doc/modules.txt
-%doc doc/PEERING
-%doc doc/queue.txt
-%doc doc/rtp-packetization.txt
-%doc doc/siptls.txt
-%doc doc/smdi.txt
-%doc doc/sms.txt
-%doc doc/speechrec.txt
-%doc doc/ss7.txt
-%doc doc/video.txt
+#doc doc/backtrace.txt
+#doc doc/callfiles.txt
+#doc doc/externalivr.txt
+#doc doc/macroexclusive.txt
+#doc doc/manager_1_1.txt
+#doc doc/modules.txt
+#doc doc/PEERING
+#doc doc/queue.txt
+#doc doc/rtp-packetization.txt
+#doc doc/siptls.txt
+#doc doc/smdi.txt
+#doc doc/sms.txt
+#doc doc/speechrec.txt
+#doc doc/ss7.txt
+#doc doc/video.txt
%{_initrddir}/asterisk
%config(noreplace) %{_sysconfdir}/sysconfig/asterisk
@@ -789,6 +785,7 @@ fi
%{_libdir}/asterisk/modules/res_security_log.so
%{_libdir}/asterisk/modules/res_smdi.so
%{_libdir}/asterisk/modules/res_speech.so
+%{_libdir}/asterisk/modules/res_srtp.so
%{_libdir}/asterisk/modules/res_stun_monitor.so
%{_libdir}/asterisk/modules/res_timing_pthread.so
%if 0%{?fedora} > 0
@@ -950,10 +947,10 @@ fi
%files devel
%defattr(-,root,root,-)
-%doc doc/CODING-GUIDELINES
-%doc doc/datastores.txt
-%doc doc/modules.txt
-%doc doc/valgrind.txt
+#doc doc/CODING-GUIDELINES
+#doc doc/datastores.txt
+#doc doc/modules.txt
+#doc doc/valgrind.txt
%dir %{_includedir}/asterisk
%dir %{_includedir}/asterisk/doxygen
@@ -980,8 +977,8 @@ fi
%files jabber
%defattr(-,root,root,-)
-%doc doc/jabber.txt
-%doc doc/jingle.txt
+#doc doc/jabber.txt
+#doc doc/jingle.txt
%attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/gtalk.conf
%attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/jabber.conf
%attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/jingle.conf
@@ -1000,7 +997,7 @@ fi
%files ldap
%defattr(-,root,root,-)
-%doc doc/ldap.txt
+#doc doc/ldap.txt
%attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_ldap.conf
%{_libdir}/asterisk/modules/res_config_ldap.so
@@ -1085,12 +1082,12 @@ fi
%files snmp
%defattr(-,root,root,-)
-%doc doc/asterisk-mib.txt
-%doc doc/digium-mib.txt
-%doc doc/snmp.txt
+#doc doc/asterisk-mib.txt
+#doc doc/digium-mib.txt
+#doc doc/snmp.txt
%attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_snmp.conf
-%{_datadir}/snmp/mibs/ASTERISK-MIB.txt
-%{_datadir}/snmp/mibs/DIGIUM-MIB.txt
+#%{_datadir}/snmp/mibs/ASTERISK-MIB.txt
+#%{_datadir}/snmp/mibs/DIGIUM-MIB.txt
%{_libdir}/asterisk/modules/res_snmp.so
%files sqlite
@@ -1110,7 +1107,7 @@ fi
%files unistim
%defattr(-,root,root,-)
-%doc doc/unistim.txt
+#doc doc/unistim.txt
%attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/unistim.conf
%{_libdir}/asterisk/modules/chan_unistim.so
@@ -1133,7 +1130,7 @@ fi
%files voicemail-odbc
%defattr(-,root,root,-)
-%doc doc/voicemail_odbc_postgresql.txt
+#doc doc/voicemail_odbc_postgresql.txt
%{_libdir}/asterisk/modules/app_directory_odbc.so
%{_libdir}/asterisk/modules/app_voicemail_odbc.so
@@ -1143,6 +1140,191 @@ fi
%{_libdir}/asterisk/modules/app_voicemail_plain.so
%changelog
+* Mon Jan 24 2011 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.2.2-2
+- Build with SRTP support
+
+* Mon Jan 24 2011 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.2.2-1
+-
+- The Asterisk Development Team has announced a release for the security issue
+- described in AST-2011-001.
+-
+- Due to a failed merge, Asterisk 1.8.2.1 which should have included the security
+- fix did not. Asterisk 1.8.2.2 contains the the changes which should have been
+- included in Asterisk 1.8.2.1.
+-
+- This releases is available for immediate download at
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases
+-
+- The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2,
+- 1.8.1.2, and 1.8.2.2 resolve an issue when forming an outgoing SIP request while
+- in pedantic mode, which can cause a stack buffer to be made to overflow if
+- supplied with carefully crafted caller ID information. The issue and resolution
+- are described in the AST-2011-001 security advisory.
+-
+- For more information about the details of this vulnerability, please read the
+- security advisory AST-2011-001, which was released at the same time as this
+- announcement.
+-
+- For a full list of changes in the current release, please see the ChangeLog:
+-
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.2
+-
+- Security advisory AST-2011-001 is available at:
+-
+- http://downloads.asterisk.org/pub/security/AST-2011-001.pdf
+
+* Mon Jan 24 2011 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.2.1-1
+-
+- The Asterisk Development Team has announced security releases for the following
+- versions of Asterisk:
+-
+- * 1.4.38.1
+- * 1.4.39.1
+- * 1.6.1.21
+- * 1.6.2.15.1
+- * 1.6.2.16.1
+- * 1.8.1.2
+- * 1.8.2.1
+-
+- These releases are available for immediate download at
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases
+-
+- The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2,
+- 1.8.1.2, and 1.8.2.1 resolve an issue when forming an outgoing SIP request while
+- in pedantic mode, which can cause a stack buffer to be made to overflow if
+- supplied with carefully crafted caller ID information. The issue and resolution
+- are described in the AST-2011-001 security advisory.
+-
+- For more information about the details of this vulnerability, please read the
+- security advisory AST-2011-001, which was released at the same time as this
+- announcement.
+-
+- For a full list of changes in the current releases, please see the ChangeLog:
+-
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.38.1
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.39.1
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.21
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.15.1
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.1
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.1.2
+- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.1
+-
+- Security advisory AST-2011-001 is available at:
+-
+- http://downloads.asterisk.org/pub/security/AST-2011-001.pdf
+
+* Mon Jan 24 2011 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.2-1
+-
+- The Asterisk Development Team has announced the release of Asterisk 1.8.2. This
+- release is available for immediate download at
+- http://downloads.asterisk.org/pub/telephony/asterisk/
+-
+- The release of Asterisk 1.8.2 resolves several issues reported by the
+- community and would have not been possible without your participation.
+- Thank you!
+-
+- The following is a sample of the issues resolved in this release:
+-
+- * 'sip notify clear-mwi' needs terminating CRLF.
+- (Closes issue #18275. Reported, patched by klaus3000)
+-
+- * Patch for deadlock from ordering issue between channel/queue locks in
+- app_queue (set_queue_variables).
+- (Closes issue #18031. Reported by rain. Patched by bbryant)
+-
+- * Fix cache of device state changes for multiple servers.
+- (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested
+- by russellb)
+-
+- * Resolve issue where channel redirect function (CLI or AMI) hangs up the call
+- instead of redirecting the call.
+- (Closes issue #18171. Reported by: SantaFox)
+- (Closes issue #18185. Reported by: kwemheuer)
+- (Closes issue #18211. Reported by: zahir_koradia)
+- (Closes issue #18230. Reported by: vmarrone)
+- (Closes issue #18299. Reported by: mbrevda)
+- (Closes issue #18322. Reported by: nerbos)
+-
+- * Fix reloading of peer when a user is requested. Prevent peer reloading from
+- causing multiple MWI subscriptions to be created when using realtime.
+- (Closes issue #18342. Reported, patched by nivek.)
+-
+- * Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0
+- so res_jabber doesn't think there is already an XMPP connection sending
+- device state. Also clean up CLI commands a bit.
+- (Closes issue #18272. Reported by klaus3000. Patched by Marquis42)
+-
+- * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
+- setting peer->cdr = NULL, set it to not post.
+- (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)
+-
+- * Fixes issue with outbound google voice calls not working. Thanks to az1234
+- and nevermind_quack for their input in helping debug the issue.
+- (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
+-
+- For a full list of changes in this release, please see the ChangeLog:
+-
+- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2
+
+* Mon Jan 24 2011 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.1.1-1
+-
+- The Asterisk Development Team has announced the release of Asterisk 1.8.1.1.
+- This release is available for immediate download at
+- http://downloads.asterisk.org/pub/telephony/asterisk/
+-
+- The release of Asterisk 1.8.1.1 resolves two issues reported by the community
+- since the release of Asterisk 1.8.1.
+-
+- * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
+- setting peer->cdr = NULL, set it to not post.
+- (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)
+-
+- * Fixes issue with outbound google voice calls not working. Thanks to az1234
+- and nevermind_quack for their input in helping debug the issue.
+- (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
+-
+- For a full list of changes in this release candidate, please see the ChangeLog:
+-
+- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1.1
+
+* Mon Jan 24 2011 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.1-1
+-
+- The Asterisk Development Team has announced the release of Asterisk 1.8.1. This
+- release is available for immediate download at
+- http://downloads.asterisk.org/pub/telephony/asterisk/
+-
+- The release of Asterisk 1.8.1 resolves several issues reported by the
+- community and would have not been possible without your participation.
+- Thank you!
+-
+- The following is a sample of the issues resolved in this release:
+-
+- * Fix issue when using directmedia. Asterisk needs to limit the codecs offered
+- to just the ones that both sides recognize, otherwise they may end up sending
+- audio that the other side doesn't understand.
+- (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)
+-
+- * Resolve issue where Party A in an analog 3-way call would continue to hear
+- ringback after party C answers.
+- (Patched by rmudgett)
+-
+- * Fix playback failure when using IAX with the timerfd module.
+- (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)
+-
+- * Fix problem with qualify option packets for realtime peers never stopping.
+- The option packets not only never stopped, but if a realtime peer was not in
+- the peer list multiple options dialogs could accumulate over time.
+- (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
+- jpeeler)
+-
+- * Fix issue where it is possible to crash Asterisk by feeding the curl engine
+- invalid data.
+- (Closes issue #18161. Reported by wdoekes. Patched by tilghman)
+-
+- For a full list of changes in this release, please see the ChangeLog:
+-
+- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
+
* Fri Oct 29 2010 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.0-3
- Rebuild for new net-snmp.
diff --git a/menuselect.makeopts b/menuselect.makeopts
index be1e8d1..e722c6b 100644
--- a/menuselect.makeopts
+++ b/menuselect.makeopts
@@ -6,14 +6,14 @@ MENUSELECT_CODECS=codec_ilbc
MENUSELECT_FORMATS=format_ilbc
MENUSELECT_FUNCS=
MENUSELECT_PBX=
-MENUSELECT_RES=res_config_sqlite res_srtp res_timing_kqueue
+MENUSELECT_RES=res_config_sqlite res_timing_kqueue
MENUSELECT_OPTS_app_voicemail=
MENUSELECT_CFLAGS=LOADABLE_MODULES
MENUSELECT_EMBED=
MENUSELECT_CORE_SOUNDS=
MENUSELECT_MOH=
MENUSELECT_EXTRA_SOUNDS=
-MENUSELECT_TESTS=test_skel test_heap test_sched test_astobj2 test_dlinklists test_acl test_aoc test_app test_ast_format_str_reduce test_devicestate test_event test_func_file test_gosub test_pbx test_stringfields test_strings test_substitution test_time test_utils test_amihooks test_locale test_logger test_security_events test_poll
+MENUSELECT_TESTS=test_skel test_heap test_sched test_astobj2 test_dlinklists test_acl test_aoc test_app test_ast_format_str_reduce test_devicestate test_event test_func_file test_gosub test_pbx test_stringfields test_strings test_substitution test_time test_utils test_amihooks test_locale test_logger test_security_events test_poll test_expr
MENUSELECT_DEPSFAILED=MENUSELECT_APPS=app_osplookup
MENUSELECT_DEPSFAILED=MENUSELECT_CHANNELS=chan_h323
MENUSELECT_DEPSFAILED=MENUSELECT_CHANNELS=chan_nbs
diff --git a/sources b/sources
index 59a5553..75c0157 100644
--- a/sources
+++ b/sources
@@ -1,2 +1,2 @@
-83203b43aaf12f36bdc953d6b04d18a4 asterisk-1.8.0.tar.gz
-8f288e34a742f8be384235040376997f asterisk-1.8.0.tar.gz.asc
+213ff45524dcd1bfc08ed526ffe38878 asterisk-1.8.2.2.tar.gz
+b50b0f4f61c0971d99dfda9db2fe4517 asterisk-1.8.2.2.tar.gz.asc
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