asterisk-voicemail-plain-18.12.1-1.el8.2 > 6 6_6 3!y덏%!E/֡c !E/֡_j !:KVJgQ8/r;p'y\ʕ:/cQ@H9+ ?E++ުQd䒍`1 v*hsڄQ$>B.. (<=+*٤\&s v$ p8`8rJi&AU5k'gm }5a+o%셝rue?5);1UɴޛuܯfDbiQU$sy}cV@k2+4 vQB7pf.KBIu ִ4t\fc/u C$jdtAhAڼXkLsG:'Bp;_JiXact/]̸59_@jR@`9`gdŘ.Y2-$ ,K;(7F@n0}˭~(*ikIw$< ̲Eo8{K.8z4li9emq>(k,-llI?æ09dfc772cd19b8408f8797498bc049d1bd6f3c7f4c99b8ec3ead599984f7899a8d8760ba8f5f35f56cb2773ba49a06ec32cbd845s$3!y덏%!E/֡c !E/֡?K[te,,c8K,;IF.!}])6CwU@FqQT0s sVfrH{x~YN>}=ձ8P8Ҕr`٪XQeLtO/91/}=q Pyu_2C_\&,d M0Q{SU/΁0% ]XJ_Hf*.:):|뢡A]ȜK1w?=Um"'fĮ*TyI Eva,:'!YF=G̐"tGƔ[  ?Z]L ٙTO5ذ́Yh^ N$|B =P#>v,W|Ok7يR"Ԍ, )[돻fF~b dH7_Kuv A<{uVsF N6Cp0ZpDŽz2zD,˴ՠX>p@ބ?td# + S  06@T ^ h |  XHP(89 :ZGHIX Y\H]\^b;defltuvw݌xݠyݴ(.pCasterisk-voicemail-plain18.12.11.el8.2Store voicemail on the local filesystemVoicemail implementation for Asterisk that stores voicemail on the local filesystem.cbuildvm-x86-20.iad2.fedoraproject.orgFedora ProjectFedora ProjectGPLv2Fedora ProjectUnspecifiedhttp://www.asterisk.org/linuxx86_64==Ac㴾c㴾c㴾c㴯c㴯1dc2b92422c7047ab7d197e639fec3b0daf42ce7c91ec68213245cb695a0d3c8701d22bf201a41a267e6022ee85f5a0424dbc9eee90fbd4690838d85ce3491a9../../../../usr/lib64/asterisk/modules/app_voicemail_plain.so../../../../usr/lib64/asterisk/modules/app_directory_plain.sorootrootrootrootrootrootrootrootrootrootasterisk-18.12.1-1.el8.2.src.rpmasterisk-voicemail-implementationasterisk-voicemail-plainasterisk-voicemail-plain(x86-64)@@@@@@@@    @asteriskasterisk-voicemaillibc.so.6()(64bit)libc.so.6(GLIBC_2.14)(64bit)libc.so.6(GLIBC_2.2.5)(64bit)libc.so.6(GLIBC_2.3)(64bit)libc.so.6(GLIBC_2.3.4)(64bit)libc.so.6(GLIBC_2.4)(64bit)libpthread.so.0()(64bit)libpthread.so.0(GLIBC_2.2.5)(64bit)rpmlib(CompressedFileNames)rpmlib(FileDigests)rpmlib(PayloadFilesHavePrefix)rpmlib(PayloadIsXz)rtld(GNU_HASH)18.12.1-1.el8.218.12.1-1.el8.23.0.4-14.6.0-14.0-15.2-1 asterisk-voicemail-imapasterisk-voicemail-odbc18.12.1-1.el8.218.12.1-1.el8.24.14.3cױ@b@bbbbbbbbbbTa@a@` @`C` @`t6@`.V`!'`@` l_^@_/@_@_@_P_P_G@_^^ϧ^I^^^&@^j$@^B@^0"@^r^@]+]]Γ@]@]@]]rJ@]9]8H@],j\h\@\y\f\R@\]I>]I-I-I-I-IH,HCH@HWH@H@H@HkmHQHO@H1kH @Gu@GGG@G@G@G߮G΋@G@G@G@GG{|Gt@Gl@GiGiGg@GcGcG_@G_@G^{G]*@GO@GB@GAzG=@G9G G m@F%@FS@F^F^F @F@FF;@F;@FF@FtF#@F@FEF@F@Fzh@FvsFvsFvsFvsFvsFu"@Fu"@Fr@FAF1F@EWE@E@E8@E7hE6@E6@E6@E2"DDD(@DWIDLDGwDGwDF&@D - 18.12.1-1.2Fedora Release Engineering - 18.12.1-1.1Michal Josef Špaček - 18.12.1-1Michal Josef Špaček - 18.11.2-1Michal Josef Špaček - 18.10.1-1Michal Josef Špaček - 18.9.0-1Michal Josef Špaček - 18.8.0-1Michal Josef Špaček - 18.7.1-1Michal Josef Špaček - 18.6.0-1Michal Josef Špaček - 18.5.1-1Michal Josef Špaček - 18.4.0-1.6Jitka Plesnikova - 18.4.0-1.5Fedora Release Engineering - 18.4.0-1.4Sahana Prasad - 18.4.0-1.3Fedora Release Engineering - 18.4.0-1.2Jitka Plesnikova - 18.4.0-1.1Jared K. Smith - 18.4.0-1Jared K Smith - 18.3.0-1Jared K. Smith - 18.2.1-1Pavel Raiskup - 18.2.0-1.2Fedora Release Engineering - 18.2.0-1.1Jared K. Smith - 18.2.0-1Jared K. Smith - 18.1.0-1Jared K. Smith - 18.0.1-2Jared K. Smith - 18.0.1-1Jared K. Smith - 18.0.0-1Josef Řídký - 17.7.0-2Jared K. Smith - 17.7.0-1Josef Řídký - 17.5.0-2.3Fedora Release Engineering - 17.5.0-2.2Jitka Plesnikova - 17.5.0-2.1Jared K. Smith - 17.5.0-0.rc1.1Jared K. Smith - 17.4.0-2Jared K. Smith - 17.4.0-1Jared K. Smith - 17.4.0-0.rc2.1Jared K. Smith - 17.4.0-0.rc1.1Jared K. Smith - 17.3.0-1Jared K. Smith - 17.2.0-1Fedora Release Engineering - 17.2.0-0.rc1.2.1Tom Callaway - 17.1.0-2Jared K. Smith - 17.1.0-1Jared K. Smith - 17.1.0-0.rc1.1Jared K. Smith - 17.0.1-1Jared K. Smith - 17.0.0-2Jared K. Smith - 17.0.0-1Jared K. Smith - 16.6.1-1Jared K. Smith - 16.6.0-1Jared K. Smith - 16.5.1-1Jared K. Smith - 16.5.0-1Fedora Release Engineering - 16.4.1-2Jared K. Smith - 16.4.1-1Jitka Plesnikova - 16.4.0-2Jared K. Smith - 16.4.0-1Jared K. Smith - 16.2.1-1Jared K. Smith - 16.2.0-1Fedora Release Engineering - 16.1.0-4Björn Esser - 16.1.0-3Björn Esser - 16.1.0-2Jared Smith - 16.1.0-1Jared Smith - 16.0.1-1Jared Smith - 16.0.0-1Jared K. Smith - 15.5.0-1Jitka Plesnikova - 15.4.1-2Jared K. Smith - 15.4.1-1Jared K. Smith - 15.4.0-1jsmith - 15.3.0-1Jared Smith - 15.2.2-2Jared Smith - 15.2.2-1Jared Smith - 15.2.1-3Jared Smith - 15.2.1-2Jared Smith - 15.2.1-1Igor Gnatenko - 15.2.0-5Fedora Release Engineering - 15.2.0-4Jared Smith - 15.2.0-3Björn Esser - 15.2.0-2Jared Smith - 15.2.0-1Jared Smith - 15.1.5-1Jared Smith - 15.1.4-2Jared Smith - 15.1.4-1Jared Smith - 15.1.3-1Jared Smith - 15.1.2-1Jared Smith - 15.1.1-1Jared Smith - 15.1.0-1Jared Smith - 15.0.0-1Jared Smith - 14.6.2-1Jared Smith - 14.6.1-6Jared Smith - 14.6.1-5Jared Smith - 14.6.1-4Jared Smith - 14.6.1-3Jared Smith - 14.6.1-1Jared Smith - 14.6.0-2Jared Smith - 14.6.0-1Fedora Release Engineering - 14.5.0-4Fedora Release Engineering - 14.5.0-3Till Maas - 14.5.0-2Jared Smith - 14.5.0-1Jitka Plesnikova - 13.11.2-1.2Fedora Release Engineering - 13.11.2-1.1Jared Smith - 13.11.2-1Jared Smith - 13.11.1-1Jitka Plesnikova - 13.9.1-1.1Jared Smith - 13.9.1-1Jared Smith - 13.7.2-2.1Michal Toman - 13.7.2-2Jared Smith - 13.7.2-1Jared Smith - 13.7.1-2Jared Smith - 13.7.1-1Fedora Release Engineering - 13.3.2-3.1Jared Smith - 13.3.2-3Robert Scheck - 13.3.2-2Fedora Release Engineering - 13.3.2-1.2Jitka Plesnikova - 13.3.2-1.1Jeffrey C. Ollie - 13.3.2-1:Jeffrey C. Ollie - 13.3.1-1:Jeffrey C. Ollie - 13.3.0-1:Jeffrey C. Ollie - 13.2.0-1:Jeffrey C. Ollie - 13.1.1-1:Jeffrey C. Ollie - 13.1.0-1Peter Robinson 13.0.2-3Tom Callaway - 13.0.2-2Jeffrey C. Ollie - 13.0.2-1Jeffrey C. Ollie - 13.0.1-1Jeffrey C. Ollie - 13.0.0-1Tom Callaway - 11.13.1-2Jeffrey C. Ollie - 11.13.1-1Jeffrey C. Ollie - 11.13.0-1Jeffrey C. Ollie - 11.12.1-1Jeffrey C. Ollie - 11.12.0-1Jeffrey C. Ollie - 11.11.0-1Jitka Plesnikova - 11.10.2-2.2Fedora Release Engineering - 11.10.2-2.1Jeffrey Ollie - 11.10.2-2:Jeffrey Ollie - 11.10.2-1:Jeffrey Ollie - 11.10.1-1:Jeffrey Ollie - 11.10.0-1:Fedora Release Engineering - 11.9.0-2.1Dennis Gilmore - 11.9.0-2Jeffrey Ollie - 11.9.0-1:Jeffrey Ollie - 11.8.1-1:Jeffrey Ollie - 11.8.0-1:Jeffrey Ollie - 11.7.0-1:Jeffrey Ollie - 11.6.1-1:Jeffrey Ollie - 11.6.0-1:Jeffrey Ollie - 11.5.1-3:Jeffrey Ollie - 11.5.1-2:Jeffrey Ollie - 11.5.1-1:Fedora Release Engineering - 11.4.0-2.2Petr Pisar - 11.4.0-2.1Rex Dieter 11.4.0-2Jeffrey Ollie - 11.4.0-1:Tom Callaway - 11.3.0-2:Jeffrey Ollie - 11.3.0-1:Jeffrey Ollie - 11.2.2-1:Jeffrey Ollie - 11.2.1-1:Jeffrey Ollie - 11.2.0-1:Jeffrey Ollie - 11.1.2-1:Jeffrey Ollie - 11.1.1-1:Jeffrey Ollie - 11.1.0-1:Jeffrey Ollie - 11.0.2-1:Dan Horák - 11.0.1-3Dennis Gilmore - 11.0.1-2Jeffrey Ollie - 11.0.1-1Jeffrey Ollie - 11.0.0-1:Jeffrey Ollie - 11.0.0-0.7.rc2:Jeffrey Ollie - 11.0.0-0.6.rc1Jeffrey Ollie - 11.0.0-0.5.beta2Jeffrey Ollie - 11.0.0-0.4.beta2Jeffrey Ollie - 10.8.0-1Dan Horák - 11.0.0-0.3.beta1Dan Horák - 10.7.1-2Jeffrey Ollie - 10.7.1-1Jeffrey Ollie - 10.7.0-1Jeffrey Ollie - 10.6.1-1Jeffrey Ollie - 10.6.0-1Jeffrey Ollie - 11.0.0-0.2.beta1Fedora Release Engineering - 10.5.2-1.2Petr Pisar - 10.5.2-1.1Jeffrey Ollie - 10.5.2-1:Petr Pisar - 10.5.1-1.1Jeffrey Ollie - 10.5.1-1Jeffrey Ollie - 10.5.0-1Petr Pisar - 10.4.2-1.1Jeffrey Ollie - 10.4.2-1Jeffrey Ollie - 10.4.1-1Jeffrey Ollie - 10.4.0-1Jeffrey Ollie - 10.3.1-1Russell Bryant - 10.3.0-1Russell Bryant - 10.2.1-1Jeffrey C. Ollie - 10.1.2-2Jeffrey C. Ollie - 10.1.2-1Jeffrey C. Ollie - 10.1.1-1Jeffrey C. Ollie - 10.1.0-1Russell Bryant - 10.0.0-2Fedora Release Engineering - 10.0.0-1.1Jeffrey C. Ollie - 10.0.0-1Jeffrey C. Ollie - 10.0.0-1Jeffrey C. Ollie - 10.0.0-0.7.rc3Jeffrey C. Ollie - 10.0.0-0.6.rc2Jeffrey C. Ollie - 10.0.0-0.5.rc1Jeffrey C. Ollie - 10.0.0-0.4.beta2Jeffrey C. Ollie - 10.0.0-0.3.beta2Jeffrey C. Ollie - 10.0.0-0.2.beta2Jeffrey C. Ollie - 10.0.0-0.1.beta1Petr Sabata - 1.8.5.0-1.2Petr Sabata - 1.8.5.0-1.1Jeffrey C. Ollie - 1.8.5.0-1Jeffrey C. Ollie - 1.8.5-0.2Jeffrey C. Ollie - 1.8.5-0.1.rc1Jeffrey C. Ollie - 1.8.5-0.1.rc1Jeffrey C. Ollie - 1.8.4.4-2Jeffrey C. Ollie - 1.8.4.4-1Jeffrey C. Ollie - 1.8.4.3-3Jeffrey C. Ollie - 1.8.4.3-2Jeffrey C. Ollie - 1.8.4.3-1Jeffrey C. Ollie - 1.8.4.2-2Marcela Mašláňová - 1.8.4.2-1.2Marcela Mašláňová - 1.8.4.2-1.1Jeffrey C. Ollie - 1.8.4.2-1:Jeffrey C. Ollie - 1.8.3.3-1Jeffrey C. Ollie - 1.8.3.2-2Jeffrey C. Ollie - 1.8.3.2-1Jeffrey C. Ollie - 1.8.3.1-1 - 1.8.3-1 - 1.8.3-0.7.rc3Jeffrey C. Ollie - 1.8.3-0.6.rc2Jeffrey C. Ollie - 1.8.3-0.5.rc2Jeffrey C. Ollie - 1.8.3-0.4.rc2Fedora Release Engineering - 1.8.3-0.3.rc2Jeffrey C. Ollie - 1.8.3-0.2.rc2Jeffrey C. Ollie - 1.8.3-0.1.rc1Jeffrey C. Ollie - 1.8.2.3-1Jeffrey C. Ollie - 1.8.2.2-2Jeffrey C. Ollie - 1.8.2.2-1Jeffrey C. Ollie - 1.8.2.1-1Jeffrey C. Ollie - 1.8.2-1Jeffrey C. Ollie - 1.8.1.1-1Jeffrey C. Ollie - 1.8.1-1Dennis Gilmore - 1.8.0-6Dennis Gilmore - 1.8.0-5Dennis Gilmore - 1.8.0-4Jeffrey C. Ollie - 1.8.0-3Jeffrey C. Ollie - 1.8.0-2Jeffrey C. Ollie - 1.8.0-1Jeffrey C. Ollie - 1.8.0-0.8.rc5:Jeffrey C. Ollie - 1.8.0-0.7.rc3Jeffrey C. Ollie - 1.8.0-0.6.rc2Jeffrey C. Ollie - 1.8.0-0.5.beta5Jeffrey C. Ollie - 1.8.0-0.4.beta4Jeffrey C. Ollie - 1.8.0-0.3.beta3Jeffrey C. Ollie - 1.8.0-0.2.beta2Jeffrey C. Ollie - 1.8.0-0.1.beta2Jeffrey C. Ollie - 1.6.2.10-1Jeffrey C. Ollie - 1.6.2.8-0.3.rc1Marcela Maslanova - 1.6.2.8-0.2.rc1Jeffrey C. Ollie - 1.6.2.7-1Jeffrey C. Ollie - 1.6.2.7-0.2.rc3Jeffrey C. Ollie - 1.6.2.7-0.1.rc2Jeffrey C. Ollie - 1.6.2.6-1Jeffrey C. Ollie - 1.6.2.6-0.1.rc2Jeffrey C. Ollie - 1.6.2.5-2Jeffrey C. Ollie - 1.6.2.5-1Jeffrey C. Ollie - 1.6.2.4-1Jeffrey C. Ollie - 1.6.2.2-1Jeffrey C. Ollie - 1.6.2.1-1Jeffrey C. Ollie - 1.6.2.1-0.1.rc1Jeffrey C. Ollie - 1.6.2.0-1Jeffrey C. Ollie - 1.6.2.0-0.16.rc8Jeffrey C. Ollie - 1.6.2.0-0.15.rc7Jeffrey C. Ollie - 1.6.2.0-0.14.rc6Jeffrey C. Ollie - 1.6.2.0-0.13.rc6Jeffrey C. Ollie - 1.6.2.0-0.12.rc6Jeffrey C. Ollie - 1.6.2.0-0.11.rc5Jeffrey C. Ollie - 1.6.2.0-0.10.rc4Jeffrey C. Ollie - 1.6.2.0-0.9.rc3Jeffrey C. Ollie - 1.6.2.0-0.8.rc3Jeffrey C. Ollie - 1.6.2.0-0.7.rc3Jeffrey C. Ollie - 1.6.2.0-0.6.rc3Jeffrey C. Ollie - 1.6.2.0-0.5.rc3Jeffrey C. Ollie - 1.6.2.0-0.4.rc3Jeffrey C. Ollie - 1.6.2.0-0.3.rc2Jeffrey C. Ollie - 1.6.2.0-0.2.rc2Jeffrey C. Ollie - 1.6.2.0-0.1.rc2Jeffrey C. Ollie - 1.6.1.6-2Jeffrey C. Ollie - 1.6.1.6-1Jeffrey C. Ollie - 1.6.1-0.26.rc1Tomas Mraz - 1.6.1-0.25.rc1Fedora Release Engineering - 1.6.1-0.24.rc1Jeffrey C. Ollie - 1.6.1-0.23.rc1Fedora Release Engineering - 1.6.1-0.22.rc1Jeffrey C. Ollie - 1.6.1-0.21.rc1Tomas Mraz - 1.6.1-0.13.beta4Jeffrey C. Ollie - 1.6.1-0.12.beta4Jeffrey C. Ollie - 1.6.1-0.10.beta4Jeffrey C. Ollie - 1.6.1-0.9.beta4Jeffrey C. Ollie - 1.6.1-0.8.beta4Jeffrey C. Ollie - 1.6.1-0.7.beta3Alex Lancaster - 1.6.1-0.6.beta2Jeffrey C. Ollie - 1.6.1-0.5.beta2Jeffrey C. Ollie - 1.6.1-0.4.beta2Jeffrey C. Ollie - 1.6.1-0.3.beta2Jeffrey C. Ollie - 1.6.1-0.2.beta2Jeffrey C. Ollie - 1.6.0.1-3Jeffrey C. Ollie - 1.6.0.1-2Jeffrey C. Ollie - 1.6.0-1- Bastien Nocera - 1.6.0-0.22.beta9Jeffrey C. Ollie - 1.6.0-0.21.beta9Jeffrey C. Ollie - 1.6.0-0.20.beta9Jeffrey C. Ollie - 1.6.0-0.19.beta9Jeffrey C. Ollie - 1.6.0-0.18.beta9Jeffrey C. Ollie - 1.6.0-0.17.beta9Jeffrey C. Ollie - 1.6.0-0.16.beta9Jeffrey C. Ollie - 1.6.0-0.15.beta9Jeffrey C. Ollie - 1.6.0-0.14.beta9Jeffrey C. Ollie - 1.6.0-0.13.beta8Jeffrey C. Ollie - 1.6.0-0.12.beta7.1Jeffrey C. Ollie - 1.6.0-0.11.beta7.1Jeffrey C. Ollie - 1.6.0-0.10.beta7Jeffrey C. Ollie - 1.6.0-0.9.beta6Jeffrey C. Ollie - 1.6.0-0.8.beta6Jeffrey C. Ollie - 1.6.0-0.6.beta6Tom "spot" Callaway - 1.6.0-0.5.beta5Jeffrey C. Ollie - 1.6.0-0.4.beta5Jeffrey C. Ollie - 1.6.0-0.3.beta4Jeffrey C. Ollie - 1.6.0-0.2.beta4Jeffrey C. Ollie - 1.6.0-0.1.beta4Jeffrey C. Ollie - 1.4.18-1Jeffrey C. Ollie - 1.4.17-1Jeffrey C. Ollie - 1.4.16.2-1Jeffrey C. Ollie - 1.4.16.1-2Jeffrey C. Ollie - 1.4.16.1-1Jeffrey C. Ollie - 1.4.16-2Jeffrey C. Ollie - 1.4.16-1Jeffrey C. Ollie - 1.4.15-7Jeffrey C. Ollie - 1.4.15-6Jeffrey C. Ollie - 1.4.15-5Jeffrey C. Ollie - 1.4.15-4Jeffrey C. Ollie - 1.4.15-3Jeffrey C. Ollie - 1.4.15-2Jeffrey C. Ollie - 1.4.15-1Jeffrey C. Ollie - 1.4.14-2Jeffrey C. Ollie - 1.4.14-1Jeffrey C. Ollie - 1.4.13-7Jeffrey C. Ollie - 1.4.13-6Jeffrey C. Ollie - 1.4.13-1Jeffrey C. Ollie - 1.4.12.1-1Jeffrey C. Ollie - 1.4.11-1Jeffrey C. Ollie - 1.4.10.1-1Jeffrey C. Ollie - 1.4.10-1Jeffrey C. Ollie - 1.4.9-7Jeffrey C. Ollie - 1.4.9-6Jeffrey C. Ollie - 1.4.9-5Jeffrey C. Ollie - 1.4.9-4Jeffrey C. Ollie - 1.4.9-3Jeffrey C. Ollie - 1.4.9-2Jeffrey C. Ollie - 1.4.9-1Jeffrey C. Ollie - 1.4.8-1Jeffrey C. Ollie - 1.4.7.1-1Jeffrey C. Ollie - 1.4.7-1Jeffrey C. Ollie - 1.4.6-4Jeffrey C. Ollie - 1.4.6-3Jeffrey C. Ollie - 1.4.6-2Jeffrey C. Ollie - 1.4.6-1Jeffrey C. Ollie - 1.4.5-10Jeffrey C. Ollie - 1.4.5-9Jeffrey C. Ollie - 1.4.5-8Jeffrey C. Ollie - 1.4.5-7Jeffrey C. Ollie - 1.4.5-6Jeffrey C. Ollie - 1.4.5-5Jeffrey C. Ollie - 1.4.5-4Jeffrey C. Ollie - 1.4.5-3Jeffrey C. Ollie - 1.4.5-1Jeffrey C. Ollie - 1.4.4-2Jeffrey C. Ollie - 1.4.4-1Jeffrey C. Ollie - 1.4.2-1Jeffrey C. Ollie - 1.4.1-2Jeffrey C. Ollie - 1.4.1-1Jeffrey C. Ollie - 1.4.0-6.beta4Jeffrey C. Ollie - 1.4.0-5.beta3Jeffrey C. Ollie - 1.4.0-4.beta3Jeffrey C. Ollie - 1.4.0-3.beta3Jeffrey C. Ollie - 1.4.0-2.beta3Jeffrey C. Ollie - 1.4.0-1.beta3Jeffrey C. Ollie - 1.4.0-0.beta2Jeffrey C. Ollie - 1.2.10-1Jeffrey C. Ollie - 1.2.9.1Jeffrey C. Ollie - 1.2.8Jeffrey C. Ollie - 1.2.7.1-6Jeffrey C. Ollie - 1.2.7.1-5Jeffrey C. Ollie - 1.2.7.1-4Jeffrey C. Ollie - 1.2.7.1-3Jeffrey C. Ollie - 1.2.7.1-2Jeffrey C. Ollie - 1.2.7-1Jeffrey C. Ollie - 1.2.6-3Jeffrey C. Ollie - 1.2.6-2Jeffrey C. Ollie - 1.2.6-1Jeffrey C. Ollie - 1.2.5-1Jeffrey C. Ollie - 1.2.4-4Jeffrey C. Ollie - 1.2.4-3Jeffrey C. Ollie - 1.2.4-2Jeffrey C. Ollie - 1.2.4-1Jeffrey C. Ollie - 1.2.3-4Jeffrey C. Ollie - 1.2.3-3Jeffrey C. Ollie - 1.2.3-2Jeffrey C. Ollie - 1.2.3-1- Rebuilt to change Python shebangs to /usr/bin/python3.6 on EPEL 8- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild- Update to upstream 18.12.1 release.- Update to upstream 18.11.2 release.- Update to upstream 18.10.1 release.- Update to upstream 18.9.0 release.- Update to upstream 18.8.0 release.- Update to upstream 18.7.1 release.- Update to upstream 18.6.0 release.- Update to upstream 18.5.1 release.- Fix build (#1977579)- Perl 5.36 rebuild- Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild- Rebuilt with OpenSSL 3.0.0- Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild- Perl 5.34 rebuild- Update to upstream 18.4.0 release for bug fixes- Update to upstream 18.3.0 release for security updates and bug fixes- Update to upstream 18.2.1 release for security updates, related to: - AST-2021-001/CVE-2020-35776: Remote crash in res_pjsip_diversion - AST-2021-002/CVE-2021-26717: Remote crash possible when negotiating T.38 - AST-2021-003/CVE-2021-26712: Remote attacker could prematurely tear down SRTP calls - AST-2021-004/CVE-2021-26714: An unsuspecting user could crash Asterisk with multiple hold/unhold requests - AST-2021-005/CVE-2021-26906: Remote Crash Vulnerability in PJSIP channel driver- rebuild for libpq ABI fix rhbz#1908268- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild- Update to upstream 18.2.0 release for security fixes, bug fixes, and features- Update to upstream 18.1.0 release for bug fixes and features- Add dependency on sox- Update to 18.0.1 release for AST-2020-001 and AST-2020-002 security fixes- Update to upstream 18.0.0 release for new features- Rebuilt for new net-snmp release- Update to upstream 17.7.0 release- Rebuilt for new net-snmp release- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild- Perl 5.32 rebuild - Add missing source files- Update to upststream 7.5.0-rc1 release for testing- app_page no longer depends on app_meetme- Update to upstream 7.4.0 release for bug fixes- Update to upstream 7.4.0-rc2- Update to upstream 7.4.0 RC 1- Update to upstream 7.3.0 release for bug fixes- Update to upstream 7.2.0 release for bug fixes- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild- rebuild for libsrtp2- Update to upstream 17.1.0 release for security and bug fixes- Update to upstream 17.1.0 pre-release for security and bug fixes- Update to upstream 17.0.1 release for AST-2019-006, AST-2019-007, AST-2019-008 security updates- Move from python2 to python3- Update to upstream 17.0.0 release for new features- Update to upstream 16.6.1 for bug fixes - Work on building in EPEL-7 and EPEL-8- Update to upstream 16.6.0 for security and bug fixes - Update to using bundled pjproject release 2.9- Update for upstream security release 16.5.1, with AST-2019-004 and AST-2019-005- Update to upstream 16.5.0 release for security and bug fixes- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild- Update to upstream 16.4.1 release for security updates AST-2019-002 and AST-2019-003 related to remote crash vulnerabilities- Perl 5.30 rebuild- Update to upstream 16.4.0 release for bug fixes- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001- Update to upstream 16.2.0 release for bug fixes- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild- Rebuilt for libcrypt.so.2 (#1666033)- Add patch to explicitly use python2 shebangs- Update to upstream 16.1.0 security release- Update to upstream 16.0.1 security release- Update to upstream 16.0.0 release- Update to upstream 15.5.0 release for security and bug fixes- Perl 5.28 rebuild- Update to upstream 15.4.1 release for AST-2018-007 and AST-2018-008 security issues- Update to upstream 15.4.0 release- Update to upstream 15.3.0 release- Update asterisk.service to wait for the network to come up- Update to upstream 15.2.2 release for security updates - This update addresses security alerts AST-2018-001 through AST-2018-006 - Upstream changelog at https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2- Verify GPG signatures on source packages- Add missing BuildRequires on gcc/gcc-c++- Update to upstream 15.2.1 release- Escape macros in %changelog- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild- Update requirements for systemd- Rebuilt for switch to libxcrypt- Update to upstream 15.2.0 release - Upstream changelog at http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0- Update to upstream 15.1.5 release for AST-2017-014/CVE-2017-17850 security issue- Require mariadb-connector-c-devel, see RHBZ#1488483- Update to upstream 15.1.4 release for AST-2017-012 security issue- Update to upstream 15.1.3 release for security issue AST-2017-013- Update to upstream 15.1.2 release- Update to upstream 15.1.1 release for AST-2017-09, AST-2017-010, and AST-2017-011 security updates- Update to upstream 15.1.0 release- Update to upstream 15.0.0 release- Update to upstream 14.6.2 release- Re-enable corosync, see RHBZ#1478089- Add dependency on unbound-devel for res_resolver_unbound- Disable corosync modules until corosync works in ppc64le again- Fix MySQL header path (due to change in mariadb-devel patckage)- Update to upstream 14.6.1 release - Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues- Add perl to BuildRequires- Update to upstream 14.6.0 release - Re-enable radius sub-packages- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild- Excludearch s390x- Update to upstream 14.5.0 release- Perl 5.26 rebuild- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild- Update to upstream 13.11.2 bug-fix release- Stop building the -radius subpackage, due to orphaned freeradius-client dependency - Update to upstream 13.11.1 security release for AST-2016-006 and AST-2016-007- Perl 5.24 rebuild- Update to upstream 13.9.1 release - Use bootstrap.sh instead of calling autoconf tools manually - Fix up shifting pjproject submodules - Fix up requires on speexdsp-devel for EPEL7 (RHBZ#1310444)- Fix alembic requirement- Do not use -m32/-m64 on MIPS- Update to upstream release 13.7.2 to fix ASTERISK-25702- Create sub-package for alembic scripts- Update to upstream 13.7.1 release for security fixes - Resolves AST-2016-001: BEAST vulnerability in HTTP server - Resolves AST-2016-002: File descriptor exhaustion in chan_sip - Resolves AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data - Full changelog at http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1 - Also build the 'radius' sub-package against freeradius-client-devel, as the radiusclient-ng project is dead- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild- Remove %defattr macro invocations, as they are no longer needed- Rebuild for libical 2.0.0- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild- Perl 5.22 rebuild- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11, - 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerability: - - * AST-2015-003: TLS Certificate Common name NULL byte exploit - - When Asterisk registers to a SIP TLS device and and verifies the server, - Asterisk will accept signed certificates that match a common name other than - the one Asterisk is expecting if the signed certificate has a common name - containing a null byte after the portion of the common name that Asterisk - expected. This potentially allows for a man in the middle attack. - - For more information about the details of this vulnerability, please read - security advisory AST-2015-003, which was released at the same time as this - announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf- The Asterisk Development Team has announced the release of Asterisk 13.3.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.3.1 resolves an issue reported by the - community and would have not been possible without your participation. - Thank you! - - The following is the issue resolved in this release: - - * --- pjsip: resolve compatibility problem with ast_sip_session - (Closes issue ASTERISK-24941. Reported by Matt Jordan) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1- The Asterisk Development Team has announced the release of Asterisk 13.3.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.3.0 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: - - New Features made in this release: - ----------------------------------- - * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a - channel (Reported by Matt Jordan) - * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation - (Reported by Dwayne Hubbard) - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid - string copy (Reported by Yura Kocyuba) - * ASTERISK-24748 - res_pjsip: If wizards explicitly configured in - sorcery.conf false ERROR messages may occur (Reported by Joshua - Colp) - * ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked - (Reported by Matt Jordan) - * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in - res_odbc (Reported by ibercom) - * ASTERISK-24479 - Enable REF_DEBUG for module references - (Reported by Corey Farrell) - * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to - fully disconnect underlying socket, leading to events being - dropped with no additional information (Reported by Matt Jordan) - * ASTERISK-24772 - ODBC error in realtime sippeers when device - unregisters under MariaDB (Reported by Richard Miller) - * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge - is destroyed by ARI during shutdown (Reported by Richard - Mudgett) - * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported - by Zane Conkle) - * ASTERISK-24015 - app_transfer fails with PJSIP channels - (Reported by Private Name) - * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk - transfer scenario. (Reported by Mark Michelson) - * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by - Niklas Larsson) - * ASTERISK-24716 - Improve pjsip log messages for presence - subscription failure (Reported by Rusty Newton) - * ASTERISK-24612 - res_pjsip: No information if a required sorcery - wizard is not loaded (Reported by Joshua Colp) - * ASTERISK-24768 - res_timing_pthread: file descriptor leak - (Reported by Matthias Urlichs) - * ASTERISK-24685 - "pjsip show version" CLI command (Reported by - Joshua Colp) - * ASTERISK-24632 - install_prereq script installs pjproject - without IPv6 support (Reported by Rusty Newton) - * ASTERISK-24085 - Documentation - We should remove or further - document the 'contact' section in pjsip.conf (Reported by Rusty - Newton) - * ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by - JoshE) - * ASTERISK-24700 - CRASH: NULL channel is being passed to - ast_bridge_transfer_attended() (Reported by Zane Conkle) - * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove - (Reported by Corey Farrell) - * ASTERISK-24799 - [patch] make fails with undefined reference to - SSLv3_client_method (Reported by Alexander Traud) - * ASTERISK-22670 - Asterisk crashes when processing ISDN AoC - Events (Reported by klaus3000) - * ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn - call (Reported by Marcel Manz) - * ASTERISK-24740 - [patch]Segmentation fault on aoc-e event - (Reported by Panos Gkikakis) - * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility - for playing back messages stored in IMAP - play_message: No - origtime (Reported by Graham Barnett) - * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc - OSX with 64 bit integers (Reported by Corey Farrell) - * ASTERISK-24796 - Codecs and bucket schema's prevent module - unload (Reported by Corey Farrell) - * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML - (Reported by Ashley Sanders) - * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring - is invalid (Reported by Rusty Newton) - * ASTERISK-24785 - 'Expires' header missing from 200 OK on - REGISTER (Reported by Ross Beer) - * ASTERISK-24677 - ARI GET variable on channel provides unhelpful - response on non-existent variable (Reported by Joshua Colp) - * ASTERISK-24797 - bridge_softmix: G.729 codec license held - (Reported by Kevin Harwell) - * ASTERISK-24812 - ARI: Creating channels through /channels - resource always uses SLIN, which results in unneeded transcoding - (Reported by Matt Jordan) - * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid - thread ID being passed to pthread_kill (Reported by JoshE) - * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime - fail (Reported by Terry Wilson) - * ASTERISK-23214 - chan_sip WARNING message 'We are requesting - SRTP for audio, but they responded without it' is ambiguous and - wrong in some cases (Reported by Rusty Newton) - * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an - error response and BYE are sent to the caller (Reported by - Makoto Dei) - * ASTERISK-18105 - most of asterisk modules are unbuildable in - cygwin environment (Reported by feyfre) - * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell) - * ASTERISK-24751 - Integer values in json payload to ARI cause - asterisk to crash (Reported by jeffrey putnam) - * ASTERISK-24838 - chan_sip: Locking inversion occurs when - building a peer causes a peer poke during request handling - (Reported by Richard Mudgett) - * ASTERISK-24825 - Caller ID not recognized using - Centrex/Distinctive dialing (Reported by Richard Mudgett) - * ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not - HAVE_PJPROJECT (Reported by Stefan Engström) - * ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers - (Reported by Kevin Harwell) - * ASTERISK-24755 - Asterisk sends unexpected early BYE to - transferrer during attended transfer when using a Stasis bridge - (Reported by John Bigelow) - * ASTERISK-24739 - [patch] - Out of files -- call fails -- - numerous files with inodes from under /usr/share/zoneinfo, - mostly posixrules (Reported by Ed Hynan) - * ASTERISK-23390 - NewExten Event with application AGI shows up - before and after AGI runs (Reported by Benjamin Keith Ford) - * ASTERISK-24786 - [patch] - Asterisk terminates when playing a - voicemail stored in LDAP (Reported by Graham Barnett) - * ASTERISK-24808 - res_config_odbc: Improper escaping of - backslashes occurs with MySQL (Reported by Javier Acosta) - * ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported - by Anatoli) - * ASTERISK-20850 - [patch]Nested functions aren't portable. - Adapting RAII_VAR to use clang/llvm blocks to get the - same/similar functionality. (Reported by Diederik de Groot) - * ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI - connection on error (Reported by Dmitriy Serov) - * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported - by Frank DiGennaro) - * ASTERISK-21038 - Bad command completion of "core set debug - channel" (Reported by Richard Kenner) - * ASTERISK-18708 - func_curl hangs channel under load (Reported by - Dave Cabot) - * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by - Atis Lezdins) - * ASTERISK-24876 - Investigate reference leaks from - tests/channels/local/local_optimize_away (Reported by Corey - Farrell) - * ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported - by Corey Farrell) - * ASTERISK-24817 - init_logger_chain: unreachable code block - (Reported by Corey Farrell) - * ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by - snuffy) - * ASTERISK-24879 - [patch]Compilation fails due to 64bit time - under OpenBSD (Reported by snuffy) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes - (Reported by Ben Merrills) - * ASTERISK-24811 - asterisk-publication sorcery object does not - use realtime (Reported by Matt Hoskins) - * ASTERISK-24790 - Reduce spurious noise in logs from voicemail - - Couldn't find mailbox %s in context (Reported by Graham Barnett) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0- The Asterisk Development Team has announced the release of Asterisk 13.2.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.2.0 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: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them - all at the same time. (Reported by Richard Mudgett) - * ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow - when using non-default sorcery wizard (Reported by Kevin - Harwell) - * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS - from JSSIP (Reported by Badalian Vyacheslav) - * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined - media streams results in 488 (Reported by Matt Jordan) - * ASTERISK-24563 - Direct Media calls within private network - sometimes get one way audio (Reported by Kevin Harwell) - * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to - race condition in accessing codec in stored ast_frame and codec - core (Reported by Matt Jordan) - * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag - enabled (Reported by Richard Mudgett) - * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is - enabled (Reported by Andreas Steinmetz) - * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly - casts char to unsigned int (Reported by Walter Doekes) - * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra - channel (Reported by Niklas Larsson) - * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is - chosen for RTP compatible channels when the DTMF mode is not - compatible (Reported by Yaniv Simhi) - * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher - level - 'Remote address is null, most likely RTP has been - stopped' (Reported by Rusty Newton) - * ASTERISK-24513 - Local channel apparently leaked in off-nominal - DTMF attended transfer (Reported by Mark Michelson) - * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present - on startup (Reported by Richard Kenner) - * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong - destination when 'sendrpid=yes' (in proxy environment) (Reported - by Karsten Wemheuer) - * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall - calls to the transferrer. (Reported by Richard Mudgett) - * ASTERISK-24376 - res_pjsip_refer: REFER request for remote - session attempts to direct channel to external_replaces - extension instead of context, without providing for the - Referred-To SIP URI (Reported by Matt Jordan) - * ASTERISK-24591 - Stasis() side of an ARI originated channel - cannot be Redirected (Reported by Kinsey Moore) - * ASTERISK-24049 - Asterisk Manager Interface: A number of list - type responses aren't using astman_send_listack (Reported by - Jonathan Rose) - * ASTERISK-24637 - Channel re-enters Stasis() when it should not - (Reported by John Bigelow) - * ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does - not function (Reported by John Kiniston) - * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT - (Reported by Kristian Høgh) - * ASTERISK-20744 - [patch] Security event logging does not work - over syslog (Reported by Michael Keuter) - * ASTERISK-24665 - Configure check required for - pjsip_get_dest_info() (Reported by Mark Michelson) - * ASTERISK-23850 - Park Application does not respect Return - Context Priority (Reported by Andrew Nagy) - * ASTERISK-23991 - [patch]asterisk.pc file contains a small error - in the CFlags returned (Reported by Diederik de Groot) - * ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown - while attempting to publish (Reported by Kevin Harwell) - * ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown - (Reported by Corey Farrell) - * ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails - on cross compilation (Reported by abelbeck) - * ASTERISK-24624 - Transfer to invalid extension results in hung - channel. (Reported by Zane Conkle) - * ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf, - Incorrect External Addresses is Used in SIP Packets When - Responding to INVITE (Reported by David Justl) - * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail - - voicemail is not deleted after review, hangup (Reported by LEI - FU) - * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects - 32-bit packages on 64-bit hosts (Reported by Ben Klang) - * ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding - to most traffic, potential deadlock (Reported by Jeff Collell) - * ASTERISK-24560 - Creating a named ARI bridge twice causes a - crash (Reported by Kinsey Moore) - * ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when - MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported - by Matt Jordan) - * ASTERISK-24640 - Registration pending stays forever after sip - reload (Reported by Max Man) - * ASTERISK-24673 - outgoing sip registers cannot be removed or - modified without doing restart (or doing module unload - chan_sip.so) (Reported by Stefan Engström) - * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor - m() option does not queue an MWI event (Reported by Gareth - Palmer) - * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis - fails to get app name (Reported by John Bigelow) - * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive - column comparison for 'defaultuser' (Reported by - HZMI8gkCvPpom0tM) - * ASTERISK-24693 - Investigate and fix memory leaks in Asterisk - (Reported by Kevin Harwell) - * ASTERISK-24626 - Voicemail passwords not being stored in ARA - (Reported by Paddy Grice) - * ASTERISK-24539 - Compile fails on OSX because of sem_timedwait - in bridge_channel.c (Reported by George Joseph) - * ASTERISK-24544 - Compile fails on OSX Yosemite because of - incorrect detection of htonll and ntohll (Reported by George - Joseph) - * ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX' - no longer displays user menus (Reported by Matt Jordan) - * ASTERISK-24721 - manager: ModuleLoad action incorrectly reports - 'module not found' during a Reload operation (Reported by Matt - Jordan) - * ASTERISK-24719 - ConfBridge recording channels get stuck when - recording started/stopped more than once (Reported by Richard - Mudgett) - * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported - by Kevin Harwell) - * ASTERISK-24728 - tcptls: Bad file descriptor error when - reloading chan_sip (Reported by Kevin Harwell) - * ASTERISK-24729 - Outbound registration not occuring on new - registrations after reload. (Reported by Richard Mudgett) - * ASTERISK-24676 - Security Vulnerability: URL request injection - in libCURL (CVE-2014-8150) (Reported by Matt Jordan) - * ASTERISK-24666 - Security Vulnerability: RTP not closed after - sip call using unsupported codec (Reported by Y Ateya) - * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL - versions (Reported by Jared Biel) - * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by - Stephan Eisvogel) - * ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson) - * ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response - is ever received (Reported by Marco Paland) - * ASTERISK-24737 - When agent not logged in, agent status shows - unavailable, queue status shows agent invalid (Reported by - Richard Mudgett) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24552 - ARI: Allow associating a channel as an - initiator of an Origination for record keeping purposes - (Reported by Matt Jordan) - * ASTERISK-24553 - ARI/AMI: Include language in standard channel - snapshot output (Reported by Matt Jordan) - * ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by - Matt Jordan) - * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for - connection-oriented transports. (Reported by Matt Jordan) - * ASTERISK-24412 - [patch]Incomplete channel originate/continue - handling with ARI (Reported by Nir Simionovich (GreenfieldTech - - Israel)) - * ASTERISK-24678 - [PATCH] Added atxfer* settings to - features.conf.sample (Reported by Niklas Larsson) - * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported - by cloos) - * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by - Dan Jenkins) - * ASTERISK-24316 - For httpd server, need option to define server - name for security purposes (Reported by Andrew Nagy) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10, - 11.15.1, 12.8.1, and 13.1.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerabilities: - - * AST-2015-001: File descriptor leak when incompatible codecs are offered - - Asterisk may be configured to only allow specific audio or - video codecs to be used when communicating with a - particular endpoint. When an endpoint sends an SDP offer - that only lists codecs not allowed by Asterisk, the offer - is rejected. However, in this case, RTP ports that are - allocated in the process are not reclaimed. - - This issue only affects the PJSIP channel driver in - Asterisk. Users of the chan_sip channel driver are not - affected. - - * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability - - CVE-2014-8150 reported an HTTP request injection - vulnerability in libcURL. Asterisk uses libcURL in its - func_curl.so module (the CURL() dialplan function), as well - as its res_config_curl.so (cURL realtime backend) modules. - - Since Asterisk may be configured to allow for user-supplied - URLs to be passed to libcURL, it is possible that an - attacker could use Asterisk as an attack vector to inject - unauthorized HTTP requests if the version of libcURL - installed on the Asterisk server is affected by - CVE-2014-8150. - - For more information about the details of these vulnerabilities, please read - security advisory AST-2015-001 and AST-2015-002, which were released at the same - time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf - * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf- The Asterisk Development Team has announced the release of Asterisk 13.1.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.1.0 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: - - New Features made in this release: - ----------------------------------- - * ASTERISK-24554 - AMI/ARI: Generate events on connected line - changes (Reported by Matt Jordan) - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling - against libsrtp-1.5.0 (Reported by Patrick Laimbock) - * ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by - Corey Farrell) - * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing - leak (Reported by Corey Farrell) - * ASTERISK-24430 - missing letter "p" in word response in - OriginateResponse event documentation (Reported by Dafi Ni) - * ASTERISK-24437 - Review implementation of ast_bridge_impart for - leaks and document proper usage (Reported by Scott Griepentrog) - * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by - Corey Farrell) - * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by - Corey Farrell) - * ASTERISK-24458 - chan_phone fails to build on big endian systems - (Reported by Tzafrir Cohen) - * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers - (Reported by Olle Johansson) - * ASTERISK-24304 - asterisk crashing randomly because of unistim - channel (Reported by dhanapathy sathya) - * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by - Nick Adams) - * ASTERISK-24462 - res_pjsip: Stale qualify statistics after - disablementation (Reported by Kevin Harwell) - * ASTERISK-24465 - audiohooks list leaks reference to formats - (Reported by Corey Farrell) - * ASTERISK-24466 - app_queue: fix a couple leaks to struct - call_queue (Reported by Corey Farrell) - * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled - (Reported by Corey Farrell) - * ASTERISK-24411 - [patch] Status of outbound registration is not - changed upon unregistering. (Reported by John Bigelow) - * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream - leaks (Reported by Corey Farrell) - * ASTERISK-24480 - res_http_websockets: Module reference decrease - below zero (Reported by Corey Farrell) - * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in - audiohook callback (Reported by Corey Farrell) - * ASTERISK-24487 - configuration: sections should be loadable as - template even when not marked (Reported by Scott Griepentrog) - * ASTERISK-20127 - [Regression] Config.c config_text_file_load() - unescapes semicolons ("\;" -> ";") turning them into comments - (corruption) on rewrite of a config file (Reported by George - Joseph) - * ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload - when DNS settings invalid (Reported by Melissa Shepherd) - * ASTERISK-24307 - Unintentional memory retention in stringfields - (Reported by Etienne Lessard) - * ASTERISK-24491 - Memory leak in res_hep (Reported by Zane - Conkle) - * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes - extra calls to ast_module_unref (Reported by Corey Farrell) - * ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when - waiting for more matching digits. (Reported by Richard Mudgett) - * ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to - queue caller (Reported by Steve Pitts) - * ASTERISK-24504 - chan_console: Fix reference leaks to pvt - (Reported by Corey Farrell) - * ASTERISK-24250 - [patch] Voicemail with multi-recipients To: - header fix (Reported by abelbeck) - * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS - length exceeds 50 (roughly) national symbols (Reported by - Dmitriy Bubnov) - * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN - revision r227276 (Reported by Xavier Hienne) - * ASTERISK-24505 - manager: http connections leak references - (Reported by Corey Farrell) - * ASTERISK-24502 - Build fails when dev-mode, dont optimize and - coverage are enabled (Reported by Corey Farrell) - * ASTERISK-24444 - PBX: Crash when generating extension for - pattern matching hint (Reported by Leandro Dardini) - * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP - packet to JSON for res_hep_rtcp and report blocks are greater - than 1 (Reported by Gregory Malsack) - * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended - transfer (Reported by Beppo Mazzucato) - * ASTERISK-24501 - ARI: Moving a channel between bridges followed - by a hangup can cause an ARI client to not receive an expected - ChannelLeftBridge event before StasisEnd (Reported by Matt - Jordan) - * ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash - (Reported by Leon Rowland) - * ASTERISK-23651 - Reloading some modules that are loaded already, - results in 'No such module' before a successful reload (Reported - by Rusty Newton) - * ASTERISK-24522 - ConfBridge: delay occurs between kicking all - endmarked users when last marked user leaves (Reported by Matt - Jordan) - * ASTERISK-15242 - transmit_refer leaks sip_refer structures - (Reported by David Woolley) - * ASTERISK-24508 - pjsip - REFER request from SNOM is rejected - with "400 bad request" - DEBUG shows "Received a REFER without a - parseable Refer-To" (Reported by Beppo Mazzucato) - * ASTERISK-24535 - stringfields: Fix regression from fix for - unintentional memory retention and another issue exposed by the - fix (Reported by Corey Farrell) - * ASTERISK-24471 - Crash - assert_fail in libc in - pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2 - (Reported by yaron nahum) - * ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces - in-dialog with invalid target causes crash (Reported by Joshua - Colp) - * ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial - module load (Reported by Matt Jordan) - * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs - allow blocked addresses through (Reported by Matt Jordan) - * ASTERISK-24542 - [patch]Failure showing codecs via 'core show - channeltype ' (Reported by snuffy) - * ASTERISK-24533 - 2 threads created per chan_sip entry (Reported - by xrobau) - * ASTERISK-24516 - [patch]Asterisk segfaults when playing back - voicemail under high concurrency with an IMAP backend (Reported - by David Duncan Ross Palmer) - * ASTERISK-24572 - [patch]App_meetme is loaded without its - defaults when the configuration file is missing (Reported by - Nuno Borges) - * ASTERISK-24573 - [patch]Out of sync conversation recording when - divided in multiple recordings (Reported by Nuno Borges) - * ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not - reliably transmitted during transfers (Reported by Matt Jordan) - * ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip - extension to another pjsip extension (Reported by Abhay Gupta) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR - property 'unanswered' (Reported by Matt Jordan) - * ASTERISK-24283 - [patch]Microseconds precision in the eventtime - column in the cel_odbc module (Reported by Etienne Lessard) - * ASTERISK-24530 - [patch] app_record stripping 1/4 second from - recordings (Reported by Ben Smithurst) - * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded - lookups (Reported by Birger "WIMPy" Harzenetter) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0- Add speexdsp as build dep as speex_echo.h has moved - rhbz 1181021- update for lua 5.3- The Asterisk Development Team has announced security releases for Certified - Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are - released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerability: - - * AST-2014-019: Remote Crash Vulnerability in WebSocket Server - - When handling a WebSocket frame the res_http_websocket module dynamically - changes the size of the memory used to allow the provided payload to fit. If a - payload length of zero was received the code would incorrectly attempt to - resize to zero. This operation would succeed and end up freeing the memory but - be treated as a failure. When the session was subsequently torn down this - memory would get freed yet again causing a crash. - - For more information about the details of this vulnerability, please read - security advisory AST-2014-019, which was released at the same time as this - announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1, - 11.14.1, 12.7.1, and 13.0.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerabilities: - - * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP - address families - - Many modules in Asterisk that service incoming IP traffic have ACL options - ("permit" and "deny") that can be used to whitelist or blacklist address - ranges. A bug has been discovered where the address family of incoming - packets is only compared to the IP address family of the first entry in the - list of access control rules. If the source IP address for an incoming - packet is not of the same address as the first ACL entry, that packet - bypasses all ACL rules. - - * AST-2014-018: Permission Escalation through DB dialplan function - - The DB dialplan function when executed from an external protocol, such as AMI, - could result in a privilege escalation. Users with a lower class authorization - in AMI can access the internal Asterisk database without the required SYSTEM - class authorization. - - In addition, the release of 11.6-cert8 and 11.14.1 resolves the following - security vulnerability: - - * AST-2014-014: High call load with ConfBridge can result in resource exhaustion - - The ConfBridge application uses an internal bridging API to implement - conference bridges. This internal API uses a state model for channels within - the conference bridge and transitions between states as different things - occur. Unload load it is possible for some state transitions to be delayed - causing the channel to transition from being hung up to waiting for media. As - the channel has been hung up remotely no further media will arrive and the - channel will stay within ConfBridge indefinitely. - - In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves - the following security vulnerability: - - * AST-2014-017: Permission Escalation via ConfBridge dialplan function and - AMI ConfbridgeStartRecord Action - - The CONFBRIDGE dialplan function when executed from an external protocol (such - as AMI) can result in a privilege escalation as certain options within that - function can affect the underlying system. Additionally, the AMI - ConfbridgeStartRecord action has options that would allow modification of the - underlying system, and does not require SYSTEM class authorization in AMI. - - Finally, the release of 12.7.1 and 13.0.1 resolves the following security - vulnerabilities: - - * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack - - The Asterisk module res_pjsip provides the ability to configure ACLs that may - be used to reject SIP requests from various hosts. However, the module - currently fails to create and apply the ACLs defined in its configuration - file on initial module load. - - * AST-2014-015: Remote crash vulnerability in PJSIP channel driver - - The chan_pjsip channel driver uses a queue approach for relating to SIP - sessions. There exists a race condition where actions may be queued to answer - a session or send ringing after a SIP session has been terminated using a - CANCEL request. The code will incorrectly assume that the SIP session is still - active and attempt to send the SIP response. The PJSIP library does not - expect the SIP session to be in the disconnected state when sending the - response and asserts. - - * AST-2014-016: Remote crash vulnerability in PJSIP channel driver - - When handling an INVITE with Replaces message the res_pjsip_refer module - incorrectly assumes that it will be operating on a channel that has just been - created. If the INVITE with Replaces message is sent in-dialog after a session - has been established this assumption will be incorrect. The res_pjsip_refer - module will then hang up a channel that is actually owned by another thread. - When this other thread attempts to use the just hung up channel it will end up - using a freed channel which will likely result in a crash. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015, - AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the same - time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf- The Asterisk Development Team is pleased to announce the release of - Asterisk 13.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - Asterisk 13 is the next major release series of Asterisk. It is a Long Term - Support (LTS) release, similar to Asterisk 11. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 13, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13 - - A short list of new features includes: - - * Asterisk security events are now provided via AMI, allowing end users to - monitor their Asterisk system in real time for security related issues. - - * Both AMI and ARI now allow external systems to control the state of a mailbox. - Using AMI actions or ARI resources, external systems can programmatically - trigger Message Waiting Indicators (MWI) on subscribed phones. This is of - particular use to those who want to build their own VoiceMail application - using ARI. - - * ARI now supports the reception/transmission of out of call text messages using - any supported channel driver/protocol stack through ARI. Users receive out of - call text messages as JSON events over the ARI websocket connection, and can - send out of call text messages using HTTP requests. - - * The PJSIP stack now supports RFC 4662 Resource Lists, allowing Asterisk to act - as a Resource List Server. This includes defining lists of presence state, - mailbox state, or lists of presence state/mailbox state; managing - subscriptions to lists; and batched delivery of NOTIFY requests to - subscribers. - - * The PJSIP stack can now be used as a means of distributing device state or - mailbox state via PUBLISH requests to other Asterisk instances. This is - analogous to Asterisk's clustering support using XMPP or Corosync; unlike - existing clustering mechanisms, using the PJSIP stack to perform the - distribution of state does not rely on another daemon or server to perform the - work. - - And much more! - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation - - A full list of all new features can also be found in the CHANGES file: - - http://svnview.digium.com/svn/asterisk/branches/13/CHANGES - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0- rebuild for new libsrtp- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28-cert2, 11.6-cert7, 1.8.31.1, - 11.13.1, 12.6.1, and 13.0.0-beta3. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerability: - - * AST-2014-011: Asterisk Susceptibility to POODLE Vulnerability - - Asterisk is susceptible to the POODLE vulnerability in two ways: - 1) The res_jabber and res_xmpp module both use SSLv3 exclusively for their - encrypted connections. - 2) The core TLS handling in Asterisk, which is used by the chan_sip channel - driver, Asterisk Manager Interface (AMI), and Asterisk HTTP Server, by - default allow a TLS connection to fallback to SSLv3. This allows for a - MITM to potentially force a connection to fallback to SSLv3, exposing it - to the POODLE vulnerability. - - These issues have been resolved in the versions released in conjunction with - this security advisory. - - For more information about the details of this vulnerability, please read - security advisory AST-2014-011, which was released at the same time as this - announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert7 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.31.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.13.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.6.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0-beta3 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-011.pdf- The Asterisk Development Team has announced the release of Asterisk 11.13.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.13.0 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: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24032 - Gentoo compilation emits warning: - "_FORTIFY_SOURCE" redefined (Reported by Kilburn) - * ASTERISK-24225 - Dial option z is broken (Reported by - dimitripietro) - * ASTERISK-24178 - [patch]fromdomainport used even if not set - (Reported by Elazar Broad) - * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload - warnings and ref leaks (Reported by Walter Doekes) - * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP - ICE candidates in SDP answer (Reported by Badalian Vyacheslav) - * ASTERISK-24019 - When a Music On Hold stream starts it restarts - at beginning of file. (Reported by Jason Richards) - * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying - if ever not able to resolve (Reported by David Herselman) - * ASTERISK-24211 - testsuite: Fix the dial_LS_options test - (Reported by Matt Jordan) - * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash - Mohod) - * ASTERISK-23577 - res_rtp_asterisk: Crash in - ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by - Jay Jideliov) - * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10) - concurrent WebRTC (avpg/encryption/icesupport) calls (Reported - by Roman Skvirsky) - * ASTERISK-24301 - Security: Out of call MESSAGE requests - processed via Message channel driver can crash Asterisk - (Reported by Matt Jordan) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24171 - [patch] Provide a manpage for the aelparse - utility (Reported by Jeremy Lainé) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.13.0- The Asterisk Development Team has announced security releases for Certified - Asterisk 11.6 and Asterisk 11 and 12. The available security releases are - released as versions 11.6-cert6, 11.12.1, and 12.5.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - Please note that the release of these versions resolves the following security - vulnerability: - - * AST-2014-010: Remote Crash when Handling Out of Call Message in Certain - Dialplan Configurations - - Additionally, the release of Asterisk 12.5.1 resolves the following security - vulnerability: - - * AST-2014-009: Remote Crash Based on Malformed SIP Subscription Requests - - Note that the crash described in AST-2014-010 can be worked around through - dialplan configuration. Given the likelihood of the issue, an advisory was - deemed to be warranted. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-009 and AST-2014-010, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert6 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.12.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.5.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-009.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-010.pdf- The Asterisk Development Team has announced the release of Asterisk 11.12.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.12.0 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: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an - empty string is a bit over zealous (Reported by Matt Jordan) - * ASTERISK-23985 - PresenceState Action response does not contain - ActionID; duplicates Message Header (Reported by Matt Jordan) - * ASTERISK-23814 - No call started after peer dialed (Reported by - Igor Goncharovsky) - * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy - should not call sip_destroy (Reported by Corey Farrell) - * ASTERISK-23818 - PBX_Lua: after asterisk startup module is - loaded, but dialplan not available (Reported by Dennis Guse) - * ASTERISK-18345 - [patch] sips connection dropped by asterisk - with a large INVITE (Reported by Stephane Chazelas) - * ASTERISK-23508 - Memory Corruption in - __ast_string_field_ptr_build_va (Reported by Arnd Schmitter) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-21178 - Improve documentation for manager command - Getvar, Setvar (Reported by Rusty Newton) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.12.0- The Asterisk Development Team has announced the release of Asterisk 11.11.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.11.0 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: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-22551 - Session timer : UAS (Asterisk) starts counting - at Invite, UAC starts counting at 200 OK. (Reported by i2045) - * ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported - by Peter Whisker) - * ASTERISK-23582 - [patch]Inconsistent column length in *odbc - (Reported by Walter Doekes) - * ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all - categories but the requested one (Reported by zvision) - * ASTERISK-23035 - ConfBridge with name longer than max (32 chars) - results in several bridges with same conf_name (Reported by - Iñaki Cívico) - * ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or - AMI when waiting to enter a conference (Reported by Matt Jordan) - * ASTERISK-23683 - #includes - wildcard character in a path more - than one directory deep - results in no config parsing on module - reload (Reported by tootai) - * ASTERISK-23827 - autoservice thread doesn't exit at shutdown - (Reported by Corey Farrell) - * ASTERISK-23609 - Security: AMI action MixMonitor allows - arbitrary programs to be run (Reported by Corey Farrell) - * ASTERISK-23673 - Security: DOS by consuming the number of - allowed HTTP connections. (Reported by Richard Mudgett) - * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite - a DEBUG level of zero (Reported by Rusty Newton) - * ASTERISK-23766 - [patch] Specify timeout for database write in - SQLite (Reported by Igor Goncharovsky) - * ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua - with Lua 5.2 or greater due to addition of goto statement - (Reported by Rusty Newton) - * ASTERISK-23818 - PBX_Lua: after asterisk startup module is - loaded, but dialplan not available (Reported by Dennis Guse) - * ASTERISK-23834 - res_rtp_asterisk debug message gives wrong - length if ICE (Reported by Richard Kenner) - * ASTERISK-23790 - [patch] - SIP From headers longer than 256 - characters result in dropped call and 'No closing bracket' - warnings. (Reported by uniken1) - * ASTERISK-23917 - res_http_websocket: Delay in client processing - large streams of data causes disconnect and stuck socket - (Reported by Matt Jordan) - * ASTERISK-23908 - [patch]When using FEC error correction, - asterisk tries considers negative sequence numbers as missing - (Reported by Torrey Searle) - * ASTERISK-23921 - refcounter.py uses excessive ram for large refs - files (Reported by Corey Farrell) - * ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against - objects that were already freed (Reported by Corey Farrell) - * ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace - between attributes (Reported by Alexander Traud) - * ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite() - (Reported by Steve Davies) - * ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking - PI) in revision 413765 breaks working environments (Reported by - Pavel Troller) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-23492 - Add option to safe_asterisk to disable - backgrounding (Reported by Walter Doekes) - * ASTERISK-22961 - [patch] DTLS-SRTP not working with SHA-256 - (Reported by Jay Jideliov) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.11.0- Perl 5.20 rebuild- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild- Drop the 389 directory server schema (1061414)- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security - releases are released as versions 1.8.15-cert7, 11.6-cert4, 1.8.28.2, 11.10.2, - and 12.3.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - These releases resolve security vulnerabilities that were previously fixed in - 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1. Unfortunately, the fix - for AST-2014-007 inadvertently introduced a regression in Asterisk's TCP and TLS - handling that prevented Asterisk from sending data over these transports. This - regression and the security vulnerabilities have been fixed in the versions - specified in this release announcement. - - The security patches for AST-2014-007 have been updated with the fix for the - regression, and are available at http://downloads.asterisk.org/pub/security - - Please note that the release of these versions resolves the following security - vulnerabilities: - - * AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe - Framework - - * AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized - Shell Access - - * AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP - Connections - - * AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-005, AST-2014-006, AST-2014-007, and AST-2014-008, - which were released with the previous versions that addressed these - vulnerabilities. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert7 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.3.2 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-005.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-006.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-007.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-008.pdf- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security - releases are released as versions 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, - and 12.3.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following issue: - - * AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP - Connections - - Establishing a TCP or TLS connection to the configured HTTP or HTTPS port - respectively in http.conf and then not sending or completing a HTTP request - will tie up a HTTP session. By doing this repeatedly until the maximum number - of open HTTP sessions is reached, legitimate requests are blocked. - - Additionally, the release of 11.6-cert3, 11.10.1, and 12.3.1 resolves the - following issue: - - * AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized - Shell Access - - Manager users can execute arbitrary shell commands with the MixMonitor manager - action. Asterisk does not require system class authorization for a manager - user to use the MixMonitor action, so any manager user who is permitted to use - manager commands can potentially execute shell commands as the user executing - the Asterisk process. - - Additionally, the release of 12.3.1 resolves the following issues: - - * AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe - Framework - - A remotely exploitable crash vulnerability exists in the PJSIP channel - driver's pub/sub framework. If an attempt is made to unsubscribe when not - currently subscribed and the endpoint's “sub_min_expiry” is set to zero, - Asterisk tries to create an expiration timer with zero seconds, which is not - allowed, so an assertion raised. - - * AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions - - When a SIP transaction timeout caused a subscription to be terminated, the - action taken by Asterisk was guaranteed to deadlock the thread on which SIP - requests are serviced. Note that this behavior could only happen on - established subscriptions, meaning that this could only be exploited if an - attacker bypassed authentication and successfully subscribed to a real - resource on the Asterisk server. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-005, AST-2014-006, AST-2014-007, and AST-2014-008, - which were released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert6 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.1 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert3 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.3.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-005.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-006.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-007.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-008.pdf- The Asterisk Development Team has announced the release of Asterisk 11.10.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.10.0 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: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-23547 - [patch] app_queue removing callers from queue - when reloading (Reported by Italo Rossi) - * ASTERISK-23559 - app_voicemail fails to load after fix to - dialplan functions (Reported by Corey Farrell) - * ASTERISK-22846 - testsuite: masquerade super test fails on all - branches (still) (Reported by Matt Jordan) - * ASTERISK-23545 - Confbridge talker detection settings - configuration load bug (Reported by John Knott) - * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think - (Reported by Walter Doekes) - * ASTERISK-23620 - Code path in app_stack fails to unlock list - (Reported by Bradley Watkins) - * ASTERISK-23616 - Big memory leak in logger.c (Reported by - ibercom) - * ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS - (Reported by Sebastian Wiedenroth) - * ASTERISK-23550 - Newer sound sets don't show up in menuselect - (Reported by Rusty Newton) - * ASTERISK-18331 - app_sms failure (Reported by David Woodhouse) - * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by - Krzysztof Chmielewski) - * ASTERISK-23605 - res_http_websocket: Race condition in shutting - down websocket causes crash (Reported by Matt Jordan) - * ASTERISK-23707 - Realtime Contacts: Apparent mismatch between - PGSQL database state and Asterisk state (Reported by Mark - Michelson) - * ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial - 'spy', if the spied-on channel makes a new call, unable to - barge. (Reported by Robert Moss) - * ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+) - (Reported by Guillaume Maudoux) - * ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported - by Guillaume Maudoux) - * ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event - for INVITE/w/replaces pickup (Reported by Walter Doekes) - * ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone - (Reported by Steve Davies) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-23649 - [patch]Support for DTLS retransmission - (Reported by NITESH BANSAL) - * ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently - available in a CLI command (Reported by Patrick Laimbock) - * ASTERISK-23754 - [patch] Use var/lib directory for log file - configured in asterisk.conf (Reported by Igor Goncharovsky) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.10.0- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild- build against gmime-devel not gmime22-devel - do not use -m64 on aarch64- The Asterisk Development Team has announced the release of Asterisk 11.9.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.9.0 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: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-22790 - check_modem_rate() may return incorrect rate - for V.27 (Reported by Paolo Compagnini) - * ASTERISK-23034 - [patch] manager Originate doesn't abort on - failed format_cap allocation (Reported by Corey Farrell) - * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in - sip.conf.sample (Reported by Eugene) - * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted - minus signs (Reported by Jeremy Lainé) - * ASTERISK-23046 - Custom CDR fields set during a GoSUB called - from app_queue are not inserted (Reported by Denis Pantsyrev) - * ASTERISK-23027 - [patch] Spelling typo "transfered" instead of - "transferred" (Reported by Jeremy Lainé) - * ASTERISK-23008 - Local channels loose CALLERID name when DAHDI - channel connects (Reported by Michael Cargile) - * ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted - request and request queue may differ - fix for locking (Reported - by adomjan) - * ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image - media offer due to invalid or unsupported syntax (Reported by - adomjan) - * ASTERISK-22861 - [patch]Specifying a null time as parameter to - GotoIfTime or ExecIfTime causes segmentation fault (Reported by - Sebastian Murray-Roberts) - * ASTERISK-17837 - extconfig.conf - Maximum Include level (1) - exceeded (Reported by pz) - * ASTERISK-22662 - Documentation fix? - queues.conf says - persistentmembers defaults to yes, it appears to lie (Reported - by Rusty Newton) - * ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot - handle selinux port restrictions (Reported by Corey Farrell) - * ASTERISK-23220 - STACK_PEEK function with no arguments causes - crash/core dump (Reported by James Sharp) - * ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload' - command multiple times on cli_aliases (Reported by Joel Vandal) - * ASTERISK-22757 - segfault in res_clialiases.so on reload when - mapping "module reload" command (Reported by Gareth Blades) - * ASTERISK-17727 - [patch] TLS doesn't get all certificate chain - (Reported by LN) - * ASTERISK-23178 - devicestate.h: device state setting functions - are documented with the wrong return values (Reported by - Jonathan Rose) - * ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value - is opposite to what's expected (Reported by Leon Roy) - * ASTERISK-23098 - [patch]possible null pointer dereference in - format.c (Reported by Marcello Ceschia) - * ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if - res_parking.so is not loaded, or if res_parking.conf has no - configuration (Reported by CJ Oster) - * ASTERISK-23069 - Custom CDR variable not recorded when set in - macro called from app_queue (Reported by Bryan Anderson) - * ASTERISK-19499 - ConfBridge MOH is not working for transferee - after attended transfer (Reported by Timo Teräs) - * ASTERISK-23261 - [patch]Output mixup in - ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) - * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic - payload change in rtp mapping in the 200 OK response (Reported - by NITESH BANSAL) - * ASTERISK-23255 - UUID included for Redhat, but missing for - Debian distros in install_prereq script (Reported by Rusty - Newton) - * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR - variables for subsequent records (Reported by zvision) - * ASTERISK-23141 - Asterisk crashes on Dial(), in - pbx_find_extension at pbx.c (Reported by Maxim) - * ASTERISK-23336 - Asterisk warning "Don't know how to indicate - condition 33 on ooh323c" on outgoing calls from H323 to SIP peer - (Reported by Alexander Semych) - * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set - to minrate=2400, then res_fax refuse to load (Reported by David - Brillert) - * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set - - probably introduced in 11.7.0 (Reported by OK) - * ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in - handle_response_invite (Reported by Walter Doekes) - * ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by - ibercom) - * ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write - (Reported by Jeremy Lainé) - * ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call - from hold (Reported by Vytis Valentinavičius) - * ASTERISK-23104 - Specifying the SetVar AMI without a Channel - cause Asterisk to crash (Reported by Joel Vandal) - * ASTERISK-21930 - [patch]WebRTC over WSS is not working. - (Reported by John) - * ASTERISK-23383 - Wrong sense test on stat return code causes - unchanged config check to break with include files. (Reported by - David Woolley) - * ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set - to yes (Reported by Alexandr Gordeev) - * ASTERISK-17523 - Qualify for static realtime peers does not work - (Reported by Maciej Krajewski) - * ASTERISK-21406 - [patch] chan_sip deadlock on monlock between - unload_module and do_monitor (Reported by Corey Farrell) - * ASTERISK-23373 - [patch]Security: Open FD exhaustion with - chan_sip Session-Timers (Reported by Corey Farrell) - * ASTERISK-23340 - Security Vulnerability: stack allocation of - cookie headers in loop allows for unauthenticated remote denial - of service attack (Reported by Matt Jordan) - * ASTERISK-23311 - Manager - MoH Stop Event fails to show up when - leaving Conference (Reported by Benjamin Keith Ford) - * ASTERISK-23420 - [patch]Memory leak in manager_add_filter - function in manager.c (Reported by Etienne Lessard) - * ASTERISK-23488 - Logic error in callerid checksum processing - (Reported by Russ Meyerriecks) - * ASTERISK-23461 - Only first user is muted when joining - confbridge with 'startmuted=yes' (Reported by Chico Manobela) - * ASTERISK-20841 - fromdomain not honored on outbound INVITE - request (Reported by Kelly Goedert) - * ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f) - at astobj2.c:120 (Reported by Jamuel Starkey) - * ASTERISK-23509 - [patch]SayNumber for Polish language tries to - play empty files for numbers divisible by 100 (Reported by - zvision) - * ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find - (Reported by JoshE) - * ASTERISK-23391 - Audit dialplan function usage of channel - variable (Reported by Corey Farrell) - * ASTERISK-23548 - POST to ARI sometimes returns no body on - success (Reported by Scott Griepentrog) - * ASTERISK-23460 - ooh323 channel stuck if call is placed directly - and gatekeeper is not available (Reported by Dmitry Melekhov) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius - against libfreeradius-client (Reported by Jeremy Lainé) - * ASTERISK-22661 - Unable to exit ChanSpy if spied channel does - not have a call in progress (Reported by Chris Hillman) - * ASTERISK-23099 - [patch] WSS: enable ast_websocket_read() - function to read the whole available data at first and then wait - for any fragmented packets (Reported by Thava Iyer)- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security - releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1, - and 12.1.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * AST-2014-001: Stack overflow in HTTP processing of Cookie headers. - - Sending a HTTP request that is handled by Asterisk with a large number of - Cookie headers could overflow the stack. - - Another vulnerability along similar lines is any HTTP request with a - ridiculous number of headers in the request could exhaust system memory. - - * AST-2014-002: chan_sip: Exit early on bad session timers request - - This change allows chan_sip to avoid creation of the channel and - consumption of associated file descriptors altogether if the inbound - request is going to be rejected anyway. - - Additionally, the release of 12.1.1 resolves the following issue: - - * AST-2014-003: res_pjsip: When handling 401/407 responses don't assume a - request will have an endpoint. - - This change removes the assumption that an outgoing request will always - have an endpoint and makes the authenticate_qualify option work once again. - - Finally, a security advisory, AST-2014-004, was released for a vulnerability - fixed in Asterisk 12.1.0. Users of Asterisk 12.0.0 are encouraged to upgrade to - 12.1.1 to resolve both vulnerabilities. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-001, AST-2014-002, AST-2014-003, and AST-2014-004, - which were released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert5 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.26.1 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.8.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.1.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-001.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-002.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-003.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-004.pdf- The Asterisk Development Team has announced the release of Asterisk 11.8.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.8.0 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: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-22544 - Italian prompt vm-options has advertisement in - it (Reported by Rusty Newton) - * ASTERISK-21383 - STUN Binding Requests Not Being Sent Back from - Asterisk to Chrome (Reported by Shaun Clark) - * ASTERISK-22478 - [patch]Can't use pound(hash) symbol for custom - DTMF menus in ConfBridge (processed as directive) (Reported by - Nicolas Tanski) - * ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for - every register message (Reported by Pawel Pierscionek) - * ASTERISK-20862 - Asterisk min and max member penalties not - honored when set with 0 (Reported by Schmooze Com) - * ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id - read (Reported by Michael Walton) - * ASTERISK-22788 - [patch] main/translate.c: access to variable f - after free in ast_translate() (Reported by Corey Farrell) - * ASTERISK-21242 - Segfault when T.38 re-invite retransmission - receives 200 OK (Reported by Ashley Winters) - * ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving - 16 bit multipart SMS with app_sms (Reported by Jan Juergens) - * ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous' - from being executed from external interfaces (Reported by Matt - Jordan) - * ASTERISK-23021 - Typos in code : "avaliable" instead of - "available" (Reported by Jeremy Lainé) - * ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported - by Gareth Palmer) - * ASTERISK-21960 - ooh323 channels stuck (Reported by Dmitry - Melekhov) - * ASTERISK-22350 - DUNDI - core dump on shutdown - segfault in - sqlite3_reset from /usr/lib/libsqlite3.so.0 (Reported by Birger - "WIMPy" Harzenetter) - * ASTERISK-22942 - [patch] - Asterisk crashed after - Set(FAXOPT(faxdetect)=t38) (Reported by adomjan) - * ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes - instead of seconds (Reported by Robert Mordec) - * ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and - core_event_dispatcher taskprocessor thread (Reported by Etienne - Lessard) - * ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping - memory when is empty (Reported by Gareth Palmer) - * ASTERISK-22871 - cel_pgsql module not loading after "reload" or - "reload cel_pgsql.so" command (Reported by Matteo) - * ASTERISK-23084 - [patch]rasterisk needlessly prints the - AST-2013-007 warning (Reported by Tzafrir Cohen) - * ASTERISK-17138 - [patch] Asterisk not re-registering after it - receives "Forbidden - wrong password on authentication" - (Reported by Rudi) - * ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support - lua 5.2 (Reported by George Joseph) - * ASTERISK-22834 - Parking by blind transfer when lot full orphans - channels (Reported by rsw686) - * ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed - SIP transfer to parking space (Reported by Tommy Thompson) - * ASTERISK-22946 - Local From tag regression with sipgate.de - (Reported by Stephan Eisvogel) - * ASTERISK-23010 - No BYE message sent when sip INVITE is received - (Reported by Ryan Tilton) - * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set - - probably introduced in 11.7.0 (Reported by OK) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-22728 - [patch] Improve Understanding Of 'Forcerport' - When Running "sip show peers" (Reported by Michael L. Young) - * ASTERISK-22659 - Make a new core and extra sounds release - (Reported by Rusty Newton) - * ASTERISK-22919 - core show channeltypes slicing (Reported by - outtolunc) - * ASTERISK-22918 - dahdi show channels slices PRI channel dnid on - output (Reported by outtolunc) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0- The Asterisk Development Team has announced the release of Asterisk 11.7.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.7.0 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: - - * --- app_confbridge: Can now set the language used for announcements - to the conference. - (Closes issue ASTERISK-19983. Reported by Jonathan White) - - * --- app_queue: Fix CLI "queue remove member" queue_log entry. - (Closes issue ASTERISK-21826. Reported by Oscar Esteve) - - * --- chan_sip: Do not increment the SDP version between 183 and 200 - responses. - (Closes issue ASTERISK-21204. Reported by NITESH BANSAL) - - * --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls - (Closes issue ASTERISK-22005. Reported by Torrey Searle) - - * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering - And Expires Header In 200ok - (Closes issue ASTERISK-22428. Reported by Ben Smithurst) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.7.0- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security - releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, - 10.12.4-digiumphones, and 11.6.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * A buffer overflow when receiving odd length 16 bit messages in app_sms. An - infinite loop could occur which would overwrite memory when a message is - received into the unpacksms16() function and the length of the message is an - odd number of bytes. - - * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk - now marks certain individual dialplan functions as 'dangerous', which will - inhibit their execution from external sources. - - A 'dangerous' function is one which results in a privilege escalation. For - example, if one were to read the channel variable SHELL(rm -rf /) Bad - Things(TM) could happen; even if the external source has only read - permissions. - - Execution from external sources may be enabled by setting 'live_dangerously' - to 'yes' in the [options] section of asterisk.conf. Although doing so is not - recommended. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2013-006 and AST-2013-007, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert4 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert3 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.24.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf- The Asterisk Development Team has announced the release of Asterisk 11.6.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.6.0 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: - - * --- Confbridge: empty conference not being torn down - (Closes issue ASTERISK-21859. Reported by Chris Gentle) - - * --- Let Queue wrap up time influence member availability - (Closes issue ASTERISK-22189. Reported by Tony Lewis) - - * --- Fix a longstanding issue with MFC-R2 configuration that - prevented users - (Closes issue ASTERISK-21117. Reported by Rafael Angulo) - - * --- chan_iax2: Fix saving the wrong expiry time in astdb. - (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) - - * --- Fix segfault for certain invalid WebSocket input. - (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0- Disable hardened build, as it's apparently causing problems loading modules.- Enable hardened build BZ#954338 - Significant clean ups- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases - are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, - and 11.5.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * A remotely exploitable crash vulnerability exists in the SIP channel driver if - an ACK with SDP is received after the channel has been terminated. The - handling code incorrectly assumes that the channel will always be present. - - * A remotely exploitable crash vulnerability exists in the SIP channel driver if - an invalid SDP is sent in a SIP request that defines media descriptions before - connection information. The handling code incorrectly attempts to reference - the socket address information even though that information has not yet been - set. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2013-004 and AST-2013-005, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert3 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.23.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.3 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.3-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.5.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2013-004.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-005.pdf - - The Asterisk Development Team has announced the release of Asterisk 11.5.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.5.0 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 Segfault In app_queue When "persistentmembers" Is Enabled - And Using Realtime - (Closes issue ASTERISK-21738. Reported by JoshE) - - * --- IAX2: fix race condition with nativebridge transfers. - (Closes issue ASTERISK-21409. Reported by alecdavis) - - * --- Fix The Payload Being Set On CN Packets And Do Not Set Marker - Bit - (Closes issue ASTERISK-21246. Reported by Peter Katzmann) - - * --- Fix One-Way Audio With auto_* NAT Settings When SIP Calls - Initiated By PBX - (Closes issue ASTERISK-21374. Reported by Michael L. Young) - - * --- chan_sip: NOTIFYs for BLF start queuing up and fail to be sent - out after retries fail - (Closes issue ASTERISK-21677. Reported by Dan Martens) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.5.0- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild- Perl 5.18 rebuild- rebuild (libical)- The Asterisk Development Team has announced the release of Asterisk 11.4.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.4.0 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 Sorting Order For Parking Lots Stored In Static Realtime - (Closes issue ASTERISK-21035. Reported by Alex Epshteyn) - - * --- Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On - A Channel - (Closes issue ASTERISK-21294. Reported by daroz) - - * --- When a session timer expires during a T.38 call, re-invite with - correct SDP - (Closes issue ASTERISK-21232. Reported by Nitesh Bansal) - - * --- Fix white noise on SRTP decryption - (Closes issue ASTERISK-21323. Reported by andrea) - - * --- Fix reload skinny with active devices. - (Closes issue ASTERISK-16610. Reported by wedhorn) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.4.0- fix build with lua 5.2- The Asterisk Development Team has announced the release of Asterisk 11.3.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.3.0 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 where chan_mobile fails to bind to first available - port - (Closes issue ASTERISK-16357. Reported by challado) - - * --- Fix Queue Log Reporting Every Call COMPLETECALLER With "h" - Extension Present - (Closes issue ASTERISK-20743. Reported by call) - - * --- Retain XMPP filters across reconnections so external modules - continue to function as expected. - (Closes issue ASTERISK-20916. Reported by kuj) - - * --- Ensure that a declined media stream is terminated with a '\r\n' - (Closes issue ASTERISK-20908. Reported by Dennis DeDonatis) - - * --- Fix pjproject compilation in certain circumstances - (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases - are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones, - and 11.2.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * A possible buffer overflow during H.264 format negotiation. The format - attribute resource for H.264 video performs an unsafe read against a media - attribute when parsing the SDP. - - This vulnerability only affected Asterisk 11. - - * A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed - in January of this year, contained a fix for Asterisk's HTTP server for a - remotely-triggered crash. While the fix prevented the crash from being - triggered, a denial of service vector still exists with that solution if an - attacker sends one or more HTTP POST requests with very large Content-Length - values. - - This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11 - - * A potential username disclosure exists in the SIP channel driver. When - authenticating a SIP request with alwaysauthreject enabled, allowguest - disabled, and autocreatepeer disabled, Asterisk discloses whether a user - exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. - - This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11 - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.20.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.2.2 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2013-001.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-002.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-003.pdf- The Asterisk Development Team has announced the release of Asterisk 11.2.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.2.1 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: - - * --- Fix astcanary startup problem due to wrong pid value from before - daemon call - (Closes issue ASTERISK-20947. Reported by Jakob Hirsch) - - * --- Update init.d scripts to handle stderr; readd splash screen for - remote consoles - (Closes issue ASTERISK-20945. Reported by Warren Selby) - - * --- Reset RTP timestamp; sequence number on SSRC change - (Closes issue ASTERISK-20906. Reported by Eelco Brolman) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.1- The Asterisk Development Team has announced the release of Asterisk 11.2.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.2.0 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: - - * --- app_meetme: Fix channels lingering when hung up under certain - conditions - (Closes issue ASTERISK-20486. Reported by Michael Cargile) - - * --- Fix stuck DTMF when bridge is broken. - (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy) - - * --- Add missing support for "who hung up" to chan_motif. - (Closes issue ASTERISK-20671. Reported by Matt Jordan) - - * --- Remove a fixed size limitation for producing SDP and change how - ICE support is disabled by default. - (Closes issue ASTERISK-20643. Reported by coopvr) - - * --- Fix chan_sip websocket payload handling - (Closes issue ASTERISK-20745. Reported by Iñaki Baz Castillo) - - * --- Fix pjproject compilation in certain circumstances - (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.0- The Asterisk Development Team has announced a security release for Asterisk 11, - Asterisk 11.1.2. This release addresses the security vulnerabilities reported in - AST-2012-014 and AST-2012-015, and replaces the previous version of Asterisk 11 - released for these security vulnerabilities. The prior release left open a - vulnerability in res_xmpp that exists only in Asterisk 11; as such, other - versions of Asterisk were resolved correctly by the previous releases. - - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following two issues: - - * Stack overflows that occur in some portions of Asterisk that manage a TCP - connection. In SIP, this is exploitable via a remote unauthenticated session; - in XMPP and HTTP connections, this is exploitable via remote authenticated - sessions. The vulnerabilities in SIP and HTTP were corrected in a prior - release of Asterisk; the vulnerability in XMPP is resolved in this release. - - * A denial of service vulnerability through exploitation of the device state - cache. Anonymous calls had the capability to create devices in Asterisk that - would never be disposed of. Handling the cachability of device states - aggregated via XMPP is handled in this release. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-014 and AST-2012-015. - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.2 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-015.pdf - - Thank you for your continued support of Asterisk - and we apologize for having - to do this twice!- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases - are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones, - and 11.1.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following two issues: - - * Stack overflows that occur in some portions of Asterisk that manage a TCP - connection. In SIP, this is exploitable via a remote unauthenticated session; - in XMPP and HTTP connections, this is exploitable via remote authenticated - sessions. - - * A denial of service vulnerability through exploitation of the device state - cache. Anonymous calls had the capability to create devices in Asterisk that - would never be disposed of. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-014 and AST-2012-015, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert10 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.19.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-015.pdf- The Asterisk Development Team has announced the release of Asterisk 11.1.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.1.0 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 execution of 'i' extension due to uninitialized variable. - (Closes issue ASTERISK-20455. Reported by Richard Miller) - - * --- Prevent resetting of NATted realtime peer address on reload. - (Closes issue ASTERISK-18203. Reported by daren ferreira) - - * --- Fix ConfBridge crash if no timing module loaded. - (Closes issue ASTERISK-19448. Reported by feyfre) - - * --- Fix the Park 'r' option when a channel parks itself. - (Closes issue ASTERISK-19382. Reported by James Stocks) - - * --- Fix an issue where outgoing calls would fail to establish audio - due to ICE negotiation failures. - (Closes issue ASTERISK-20554. Reported by mmichelson) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0- The Asterisk Development Team has announced the release of Asterisk 11.0.2. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.0.2 resolves an issue reported by the - community and would have not been possible without your participation. - Thank you! - - The following is the issue resolved in this release: - - * --- chan_local: Fix local_pvt ref leak in local_devicestate(). - (Closes issue ASTERISK-20769. Reported by rmudgett) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.2- simplify LDFLAGS setting- clean up things to allow building on arm arches- The Asterisk Development Team has announced the release of Asterisk 11.0.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.0.1 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: - - * --- chan_sip: Fix a bug causing SIP reloads to remove all entries - from the registry - (Closes issue ASTERISK-20611. Reported by Alisher) - - * --- confbridge: Fix a bug which made conferences not record with - AMI/CLI commands - (Closes issue ASTERISK-20601. Reported by Vilius) - - * --- Fix an issue with res_http_websocket where the chan_sip - WebSocket handler could not be registered. - (Closes issue ASTERISK-20631. Reported by danjenkins) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1- The Asterisk Development Team is pleased to announce the release of - Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - Asterisk 11 is the next major release series of Asterisk. It is a Long Term - Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for DTLS-SRTP in chan_sip. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0- The Asterisk Development Team has announced the second release candidate of - Asterisk 11.0.0. This release candidate is available for immediate - download at http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.0.0-rc2 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: - - * --- Fix an issue where outgoing calls would fail to establish audio - due to ICE negotiation failures. - (Closes issue ASTERISK-20554. Reported by mmichelson) - - * --- Ensure Asterisk fails TCP/TLS SIP calls when certificate - checking fails - (Closes issue ASTERISK-20559. Reported by kmoore) - - * --- Don't make chan_sip export global symbols. - (Closes issue ASTERISK-20545. Reported by kmoore) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2- The Asterisk Development Team is pleased to announce the first release candidate - of Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - All interested users of Asterisk are encouraged to participate in the - Asterisk 11 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - All Asterisk users are invited to participate in the #asterisk-testing channel - on IRC to work together in testing the many parts of Asterisk. - - Asterisk 11 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for DTLS-SRTP in chan_sip. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1- Don't forget format_ilbc module- The Asterisk Development Team is pleased to announce the second beta release of - Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - All interested users of Asterisk are encouraged to participate in the - Asterisk 11 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - All Asterisk users are invited to participate in the #asterisk-testing channel - on IRC to work together in testing the many parts of Asterisk. - - Asterisk 11 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for DTLS-SRTP in chan_sip. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta2- The Asterisk Development Team has announced the release of Asterisk 10.8.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.8.0 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: - - * --- AST-2012-012: Resolve AMI User Unauthorized Shell Access through - ExternalIVR - (Closes issue ASTERISK-20132. Reported by Zubair Ashraf of IBM X-Force Research) - - * --- AST-2012-013: Resolve ACL rules being ignored during calls by - some IAX2 peers - (Closes issue ASTERISK-20186. Reported by Alan Frisch) - - * --- Handle extremely out of order RFC 2833 DTMF - (Closes issue ASTERISK-18404. Reported by Stephane Chazelas) - - * --- Resolve severe memory leak in CEL logging modules. - (Closes issue AST-916. Reported by Thomas Arimont) - - * --- Only re-create an SRTP session when needed - (Issue ASTERISK-20194. Reported by Nicolo Mazzon) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.8.0- fix build on s390- fix build on s390- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are - released as versions 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones - resolve the following two issues: - - * A permission escalation vulnerability in Asterisk Manager Interface. This - would potentially allow remote authenticated users the ability to execute - commands on the system shell with the privileges of the user running the - Asterisk application. Please note that the README-SERIOUSLY.bestpractices.txt - file delivered with Asterisk has been updated due to this and other related - vulnerabilities fixed in previous versions of Asterisk. - - * When an IAX2 call is made using the credentials of a peer defined in a - dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that - peer are not applied to the call attempt. This allows for a remote attacker - who is aware of a peer's credentials to bypass the ACL rules set for that - peer. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-012 and AST-2012-013, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert7 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.15.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1-digiumphones - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-012.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-013.pdf- The Asterisk Development Team has announced the release of Asterisk 10.7.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.7.0 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 deadlock potential with ast_set_hangupsource() calls. - (Closes issue ASTERISK-19801. Reported by Alec Davis) - - * --- Fix request routing issue when outboundproxy is used. - (Closes issue ASTERISK-20008. Reported by Marcus Hunger) - - * --- Set the Caller ID "tag" on peers even if remote party - information is present. - (Closes issue ASTERISK-19859. Reported by Thomas Arimont) - - * --- Fix NULL pointer segfault in ast_sockaddr_parse() - (Closes issue ASTERISK-20006. Reported by Michael L. Young) - - * --- Do not perform install on existing directories - (Closes issue ASTERISK-19492. Reported by Karl Fife) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.7.0- The Asterisk Development Team has announced the release of Asterisk 10.6.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.6.1 resolves an issue reported by the - community and would have not been possible without your participation. - Thank you! - - The following is the issue resolved in this release: - - * --- Remove a superfluous and dangerous freeing of an SSL_CTX. - (Closes issue ASTERISK-20074. Reported by Trevor Helmsley) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.1- The Asterisk Development Team has announced the release of Asterisk 10.6.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.6.0 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: - - * --- format_mp3: Fix a possible crash in mp3_read(). - (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk) - - * --- Fix local channel chains optimizing themselves out of a call. - (Closes issue ASTERISK-16711. Reported by Alec Davis) - - * --- Re-add LastMsgsSent value for SIP peers - (Closes issue ASTERISK-17866. Reported by Steve Davies) - - * --- Prevent sip_pvt refleak when an ast_channel outlasts its - corresponding sip_pvt. - (Closes issue ASTERISK-19425. Reported by David Cunningham) - - * --- Send more accurate identification information in dialog-info SIP - NOTIFYs. - (Closes issue ASTERISK-16735. Reported by Maciej Krajewski) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.0- The Asterisk Development Team is pleased to announce the first beta release of - Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - All interested users of Asterisk are encouraged to participate in the - Asterisk 11 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - All Asterisk users are invited to participate in the #asterisk-testing channel - on IRC to work together in testing the many parts of Asterisk. - - Asterisk 11 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the caller/callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta1- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild- Perl 5.16 rebuild- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are - released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones - resolve the following two issues: - - * If Asterisk sends a re-invite and an endpoint responds to the re-invite with - a provisional response but never sends a final response, then the SIP dialog - structure is never freed and the RTP ports for the call are never released. If - an attacker has the ability to place a call, they could create a denial of - service by using all available RTP ports. - - * If a single voicemail account is manipulated by two parties simultaneously, - a condition can occur where memory is freed twice causing a crash. - - These issues and their resolution are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-010 and AST-2012-011, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.13.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2-digiumphones - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-010.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-011.pdf- Perl 5.16 rebuild- The Asterisk Development Team has announced a security release for Asterisk 10. - This security release is released as version 10.5.1. - - The release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 10.5.1 resolves the following issue: - - * A remotely exploitable crash vulnerability was found in the Skinny (SCCP) - Channel driver. When an SCCP client sends an Off Hook message, followed by - a Key Pad Button Message, a structure that was previously set to NULL is - dereferenced. This allows remote authenticated connections the ability to - cause a crash in the server, denying services to legitimate users. - - This issue and its resolution is described in the security advisory. - - For more information about the details of this vulnerability, please read - security advisory AST-2012-009, 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-10.5.1 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-009.pdf- The Asterisk Development Team has announced the release of Asterisk 10.5.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.5.0 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: - - * --- Turn off warning message when bind address is set to any. - (Closes issue ASTERISK-19456. Reported by Michael L. Young) - - * --- Prevent overflow in calculation in ast_tvdiff_ms on 32-bit - machines - (Closes issue ASTERISK-19727. Reported by Ben Klang) - - * --- Make DAHDISendCallreroutingFacility wait 5 seconds for a reply - before disconnecting the call. - (Closes issue ASTERISK-19708. Reported by mehdi Shirazi) - - * --- Fix recalled party B feature flags for a failed DTMF atxfer. - (Closes issue ASTERISK-19383. Reported by lgfsantos) - - * --- Fix DTMF atxfer running h exten after the wrong bridge ends. - (Closes issue ASTERISK-19717. Reported by Mario) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.5.0- Perl 5.16 rebuild- The Asterisk Development Team has announced the release of Asterisk 10.4.2. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.4.2 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: - - * --- Resolve crash in subscribing for MWI notifications - (Closes issue ASTERISK-19827. Reported by B. R) - - * --- Fix crash in ConfBridge when user announcement is played for - more than 2 users - (Closes issue ASTERISK-19899. Reported by Florian Gilcher) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.2- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are - released as versions 1.8.11-cert2, 1.8.12.1, and 10.4.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.11-cert2, 1.8.12.1, and 10.4.1 resolve the following - two issues: - - * A remotely exploitable crash vulnerability exists in the IAX2 channel - driver if an established call is placed on hold without a suggested music - class. Asterisk will attempt to use an invalid pointer to the music - on hold class name, potentially causing a crash. - - * A remotely exploitable crash vulnerability was found in the Skinny (SCCP) - Channel driver. When an SCCP client closes its connection to the server, - a pointer in a structure is set to NULL. If the client was not in the - on-hook state at the time the connection was closed, this pointer is later - dereferenced. This allows remote authenticated connections the ability to - cause a crash in the server, denying services to legitimate users. - - These issues and their resolution are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-007 and AST-2012-008, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.12.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.4.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-007.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-008.pdf- The Asterisk Development Team has announced the release of Asterisk 10.4.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.4.0 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: - - * --- Prevent chanspy from binding to zombie channels - (Closes issue ASTERISK-19493. Reported by lvl) - - * --- Fix Dial m and r options and forked calls generating warnings - for voice frames. - (Closes issue ASTERISK-16901. Reported by Chris Gentle) - - * --- Remove ISDN hold restriction for non-bridged calls. - (Closes issue ASTERISK-19388. Reported by Birger Harzenetter) - - * --- Fix copying of CDR(accountcode) to local channels. - (Closes issue ASTERISK-19384. Reported by jamicque) - - * --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors - (Closes issue ASTERISK-19303. Reported by Jon Tsiros) - - * --- Eliminate double close of file descriptor in manager.c - (Closes issue ASTERISK-18453. Reported by Jaco Kroon) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.0- The Asterisk Development Team has announced security releases for Asterisk 1.6.2, - 1.8, and 10. The available security releases are released as versions 1.6.2.24, - 1.8.11.1, and 10.3.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.6.2.24, 1.8.11.1, and 10.3.1 resolve the following two - issues: - - * A permission escalation vulnerability in Asterisk Manager Interface. This - would potentially allow remote authenticated users the ability to execute - commands on the system shell with the privileges of the user running the - Asterisk application. - - * A heap overflow vulnerability in the Skinny Channel driver. The keypad - button message event failed to check the length of a fixed length buffer - before appending a received digit to the end of that buffer. A remote - authenticated user could send sufficient keypad button message events that the - buffer would be overrun. - - In addition, the release of Asterisk 1.8.11.1 and 10.3.1 resolve the following - issue: - - * A remote crash vulnerability in the SIP channel driver when processing UPDATE - requests. If a SIP UPDATE request was received indicating a connected line - update after a channel was terminated but before the final destruction of the - associated SIP dialog, Asterisk would attempt a connected line update on a - non-existing channel, causing a crash. - - These issues and their resolution are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-004, AST-2012-005, and AST-2012-006, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.24 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.11.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.3.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-004.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-005.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-006.pdf- Update to 10.3.0- Update to 10.2.1 from upstream. - Fix remote stack overflow in app_milliwatt. - Fix remote stack overflow, including possible code injection, in HTTP digest authentication handling. - Disable asterisk-corosync package, as it doesn't build right now. - Resolves: rhbz#804045, rhbz#804038, rhbz#804042- * Add patch extracted from upstream to build with Corosync since - OpenAIS is no longer available. - * Add PrivateTmp=true to systemd service file (#782478) - * Add some macros to make it easier to build with fewer dependencies - (with corresponding less functionality) (#787389) - * Add isa macros in a few places plus a few other changes to make it - easier to cross-compile. (#787779)- The Asterisk Development Team has announced the release of Asterisk 10.1.2. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.1.2 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: - - * --- Fix SIP INFO DTMF handling for non-numeric codes --- - (Closes issue ASTERISK-19290. Reported by: Ira Emus) - - * --- Fix crash in ParkAndAnnounce --- - (Closes issue ASTERISK-19311. Reported-by: tootai) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.2- The Asterisk Development Team has announced the release of Asterisk 10.1.1. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.1.1 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: - - * --- Fixes deadlocks occuring in chan_agent --- - (Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso) - - * --- Ensure entering T.38 passthrough does not cause an infinite loop --- - (Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.1- The Asterisk Development Team is pleased to announce the release of - Asterisk 10.1.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.1.0 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: - - * AST-2012-001: prevent crash when an SDP offer - is received with an encrypted video stream when support for video - is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) - Reported by: Catalin Sanda - - * Allow playback of formats that don't support seeking. ast_streamfile - previously did unconditional seeking on files that broke playback of - formats that don't support that functionality. This patch avoids the - seek that was causing the problem. - (closes issue ASTERISK-18994) Patched by: Timo Teras - - * Add pjmedia probation concepts to res_rtp_asterisk's learning mode. In - order to better handle RTP sources with strictrtp enabled (which is the - default setting in 10) using the learning mode to figure out new sources - when they change is handled by checking for a number of consecutive (by - sequence number) packets received to an rtp struct based on a new - configurable value called 'probation'. Also, during learning mode instead - of liberally accepting all packets received, we now reject packets until a - clear source has been determined. - - * Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing - to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop - causes the loop to exit prematurely. This causes a variety of negative side - effects, depending on when the loop exits. This patch handles the frame by - essentially swallowing the frame in the local loop, as the current channel - drivers expect the RTP bridge to handle the frame, and, in the case of the - local bridge loop, no additional action is necessary. - (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested - by: Matt Jordan - - * Fix timing source dependency issues with MOH. Prior to this patch, - res_musiconhold existed at the same module priority level as the timing - sources that it depends on. This would cause a problem when music on - hold was reloaded, as the timing source could be changed after - res_musiconhold was processed. This patch adds a new module priority - level, AST_MODPRI_TIMING, that the various timing modules are now loaded - at. This now occurs before loading other resource modules, such - that the timing source is guaranteed to be set prior to resolving - the timing source dependencies. - (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, - Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont - Patched by elguero - - * Fix RTP reference leak. If a blind transfer were initiated using a - REFER without a prior reINVITE to place the call on hold, AND if Asterisk - were sending RTCP reports, then there was a reference leak for the - RTP instance of the transferrer. - (closes issue ASTERISK-19192) Reported by: Tyuta Vitali - - * Fix blind transfers from failing if an 'h' extension - is present. This prevents the 'h' extension from being run on the - transferee channel when it is transferred via a native transfer - mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported - by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by - Mark Michelson (license 5049) - - * Restore call progress code for analog ports. Extracting sig_analog - from chan_dahdi lost call progress detection functionality. Fix - analog ports from considering a call answered immediately after - dialing has completed if the callprogress option is enabled. - (closes issue ASTERISK-18841) - Reported by: Richard Miller Patched by Richard Miller - - * Fix regression that 'rtp/rtcp set debup ip' only works when a port - was also specified. - (closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by: - Walter Doekes - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.0- Remove asterisk-ais. OpenAIS was removed from Fedora.- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild- Don't build API docs as the build never finishes- The Asterisk Development Team is proud to announce the release of - Asterisk 10.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - With the release of the Asterisk 10 branch, the preceding '1.' has been removed - from the version number per the blog post available at - - - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ - - The release of Asterisk 10 would not have been possible without the support and - contributions of the community. - - You can find an overview of the work involved with the 10.0.0 release in the - summary: - - http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary.txt - - A short list of available features includes: - - * T.38 gateway functionality has been added to res_fax. - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - * Support for defining hints has been added to pbx_lua. - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES - - Also, when upgrading a system between major versions, it is imperative that you - read and understand the contents of the UPGRADE.txt file, which is located at: - - http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt- The Asterisk Development Team has announced the third release candidate of - Asterisk 10.0.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.0.0-rc3 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 candidate: - - * Add ASTSBINDIR to the list of configurable paths - - This patch also makes astdb2sqlite3 and astcanary use the configured - directory instead of relying on $PATH. - - * Don't crash on INFO automon request with no channel - - AST-2011-014. When automon was enabled in features.conf, it was possible - to crash Asterisk by sending an INFO request if no channel had been - created yet. - - * Fixed crash from orphaned MWI subscriptions in chan_sip - - This patch resolves the issue where MWI subscriptions are orphaned - by subsequent SIP SUBSCRIBE messages. - - * Fix a change in behavior in 'database show' from 1.8. - - In 1.8 and previous versions, one could use any fullword portion of - the key name, including the full key, to obtain the record. Until this - patch, this did not work for the full key. - - * Default to nat=yes; warn when nat in general and peer differ - - AST-2011-013. It is possible to enumerate SIP usernames when the general and - user/peer nat settings differ in whether to respond to the port a request is - sent from or the port listed for responses in the Via header. In 1.4 and - 1.6.2, this would mean if one setting was nat=yes or nat=route and the other - was either nat=no or nat=never. In 1.8 and 10, this would mean when one - was nat=force_rport and the other was nat=no. - - In order to address this problem, it was decided to switch the default - behavior to nat=yes/force_rport as it is the most commonly used option - and to strongly discourage setting nat per-peer/user when at all - possible. - - * Fixed SendMessage stripping extension from To: header in SIP MESSAGE - - When using the MessageSend application to send a SIP MESSAGE to a - non-peer, chan_sip stripped off the extension and failed to add it back - to the sip_pvt structure before transmitting. This patch adds the full - URI passed in from the message core to the sip_pvt structure. - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc3- The Asterisk Development Team has announced the second release candidate of - Asterisk 10.0.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.0.0-rc2 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 candidate: - - * Ensure that a null vmexten does not cause a segfault - - * Fix issue with ConfBridge participants hanging up during DTMF feature - menu usage getting stuck in conference forever - (closes issue ASTERISK-18829) - Reported by: zvision - - * Fix app_macro.c MODULEINFO section termination - (closes issue ASTERISK-18848) - Reported by: Tony Mountifield - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc2- The Asterisk Development Team is pleased to announce the first release candidate - of Asterisk 10.0.0. This release candidate is available for immediate download - at http://downloads.asterisk.org/pub/telephony/asterisk/ - - All Asterisk users are encouraged to participate in the Asterisk 10 testing - process. Please report any issues found to the issue tracker, - https://issues.asterisk.org/jira. It is also very useful to see successful test - reports. Please post those to the asterisk-dev mailing list. - - All Asterisk users are invited to participate in the #asterisk-testing - channel on IRC to work together in testing the many parts of Asterisk. - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more - information about support time lines for Asterisk releases, see the Asterisk - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - A short list of features includes: - - * T.38 gateway functionality has been added to res_fax. - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - (More information available at - https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 ) - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - * Support for defining hints has been added to pbx_lua. - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/10/CHANGES - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-rc1- Add patch from upstream SVN to fix AST-2011-012- Patch cleanup day- The Asterisk Development Team is pleased to announce the second beta release of - Asterisk 10.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - With the release of the Asterisk 10 branch, the preceding '1.' has been removed - from the version number per the blog post available at - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ - - All interested users of Asterisk are encouraged to participate in the - Asterisk 10 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - - All Asterisk users are invited to participate in the #asterisk-testing - channel on IRC to work together in testing the many parts of Asterisk. - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more - information about support time lines for Asterisk releases, see the Asterisk - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - A short list of features includes: - - * T.38 gateway functionality has been added to res_fax. - - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - - * Support for defining hints has been added to pbx_lua. - - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/10/CHANGES - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta2- - The Asterisk Development Team is pleased to announce the first beta release of - Asterisk 10.0.0-beta1. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - With the release of the Asterisk 10 branch, the preceding '1.' has been removed - from the version number per the blog post available at - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ - - All interested users of Asterisk are encouraged to participate in the - Asterisk 10 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - - All Asterisk users are invited to participate in the #asterisk-testing - channel on IRC to work together in testing the many parts of Asterisk. - Additionally users can make use of the RPM and DEB packages now being built for - all Asterisk releases. More information available at - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more - information about support time lines for Asterisk releases, see the Asterisk - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - A short list of included features includes: - - * T.38 gateway functionality has been added to res_fax. - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - * Support for defining hints has been added to pbx_lua. - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/10/CHANGES?view=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta1- Perl mass rebuild- Perl mass rebuild- The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.5.0 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 Deadlock with attended transfer of SIP call - (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, - cmaj) - - * Fixes thread blocking issue in the sip TCP/TLS implementation. - (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, - rossbeer, kowalma, Freddi_Fonet) - - * Be more tolerant of what URI we accept for call completion PUBLISH requests. - (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) - - * Fix a nasty chanspy bug which was causing a channel leak every time a spied on - channel made a call. - (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) - - * This patch fixes a bug with MeetMe behavior where the 'P' option for always - prompting for a pin is ignored for the first caller. - (Closes issue #18070. Reported by mav3rick. Patched by bbryant) - - * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If - the call that the dialplan started an AGI script for is hungup while the AGI - script is in the middle of a command then the AGI script is not notified of - the hangup. - (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) - - * Resolve issue where leaving a voicemail, the MWI message is never sent. The - same thing happens when checking a voicemail and marking it as read. - (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard - Mudgett) - - * Resolve issue where wait for leader with Music On Hold allows crosstalk - between participants. Parenthesis in the wrong position. Regression from issue - #14365 when expanding conference flags to use 64 bits. - (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5.0- Rebuild for net-snmp 5.7- Fix systemd dependencies in EL6 and F15- The Asterisk Development Team has announced the first release candidate of - Asterisk 1.8.5. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.5-rc1 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 candidate: - - * Fix Deadlock with attended transfer of SIP call - (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, - cmaj) - - * Fixes thread blocking issue in the sip TCP/TLS implementation. - (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, - rossbeer, kowalma, Freddi_Fonet) - - * Be more tolerant of what URI we accept for call completion PUBLISH requests. - (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) - - * Fix a nasty chanspy bug which was causing a channel leak every time a spied on - channel made a call. - (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) - - * This patch fixes a bug with MeetMe behavior where the 'P' option for always - prompting for a pin is ignored for the first caller. - (Closes issue #18070. Reported by mav3rick. Patched by bbryant) - - * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If - the call that the dialplan started an AGI script for is hungup while the AGI - script is in the middle of a command then the AGI script is not notified of - the hangup. - (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) - - * Resolve issue where leaving a voicemail, the MWI message is never sent. The - same thing happens when checking a voicemail and marking it as read. - (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard - Mudgett) - - * Resolve issue where wait for leader with Music On Hold allows crosstalk - between participants. Parenthesis in the wrong position. Regression from issue - #14365 when expanding conference flags to use 64 bits. - (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett) - - * Fix timerfd locking issue. - (Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1- Fedora Directory Server -> 389 Directory Server- The Asterisk Development Team has announced the release of Asterisk - versions 1.4.41.2, 1.6.2.18.2, and 1.8.4.4, which are security - releases. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the - following issue: - - AST-2011-011: Asterisk may respond differently to SIP requests from an - invalid SIP user than it does to a user configured on the system, even - when the alwaysauthreject option is set in the configuration. This can - leak information about what SIP users are valid on the Asterisk - system. - - For more information about the details of this vulnerability, please - read the security advisory AST-2011-011, 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-1.4.41.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.4 - - Security advisory AST-2011-011 is available at: - - http://downloads.asterisk.org/pub/security/AST-2011-011.pdf- Don't forget stereorize- Move /var/run/asterisk to /run/asterisk - Add comments to systemd service file on how to mimic safe_asterisk functionality - Build more of the optional binaries - Install the tmpfiles.d configuration on Fedora 15- The Asterisk Development Team has announced the release of Asterisk versions - 1.4.41.1, 1.6.2.18.1, and 1.8.4.3, which are security releases. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.4.41.1, 1.6.2.18, and 1.8.4.3 resolves several issues - as outlined below: - - * AST-2011-008: If a remote user sends a SIP packet containing a null, - Asterisk assumes available data extends past the null to the - end of the packet when the buffer is actually truncated when - copied. This causes SIP header parsing to modify data past - the end of the buffer altering unrelated memory structures. - This vulnerability does not affect TCP/TLS connections. - -- Resolved in 1.6.2.18.1 and 1.8.4.3 - - * AST-2011-009: A remote user sending a SIP packet containing a Contact header - with a missing left angle bracket (<) causes Asterisk to - access a null pointer. - -- Resolved in 1.8.4.3 - - * AST-2011-010: A memory address was inadvertently transmitted over the - network via IAX2 via an option control frame and the remote party would try - to access it. - -- Resolved in 1.4.41.1, 1.6.2.18.1, and 1.8.4.3 - - The issues and resolutions are described in the AST-2011-008, AST-2011-009, and - AST-2011-010 security advisories. - - For more information about the details of these vulnerabilities, please read - the security advisories AST-2011-008, AST-2011-009, and AST-2011-010, which were - 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-1.4.41.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.3 - - Security advisories AST-2011-008, AST-2011-009, and AST-2011-010 are available - at: - - http://downloads.asterisk.org/pub/security/AST-2011-008.pdf - http://downloads.asterisk.org/pub/security/AST-2011-009.pdf - http://downloads.asterisk.org/pub/security/AST-2011-010.pdf- Convert to systemd- Perl mass rebuild- Perl 5.14 mass rebuild- - The Asterisk Development Team has announced the release of Asterisk - version 1.8.4.2, which is a security release for Asterisk 1.8. - - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.4.2 resolves an issue with SIP URI - parsing which can lead to a remotely exploitable crash: - - Remote Crash Vulnerability in SIP channel driver (AST-2011-007) - - The issue and resolution is described in the AST-2011-007 security - advisory. - - For more information about the details of this vulnerability, please - read the security advisory AST-2011-007, 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-1.8.4.2 - - Security advisory AST-2011-007 is available at: - - http://downloads.asterisk.org/pub/security/AST-2011-007.pdf - - The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.4.1 resolves several issues reported by the - community. Without your help this release would not have been possible. - Thank you! - - Below is a list of issues resolved in this release: - - * Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix) - (Closes issue #18951. Reported by jmls. Patched by wdoekes) - - * Resolve a change in IPv6 header parsing due to the Cisco phone fix issue. - This issue was found and reported by the Asterisk test suite. - (Closes issue #18951. Patched by mnicholson) - - * Resolve potential crash when using SIP TLS support. - (Closes issue #19192. Reported by stknob. Patched by Chainsaw. Tested by - vois, Chainsaw) - - * Improve reliability when using SIP TLS. - (Closes issue #19182. Reported by st. Patched by mnicholson) - - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1 - The Asterisk Development Team has announced the release of Asterisk 1.8.4. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.4 resolves several issues reported by the community. - Without your help this release would not have been possible. Thank you! - - Below is a sample of the issues resolved in this release: - - * Use SSLv23_client_method instead of old SSLv2 only. - (Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell - and chazzam. - - * Resolve crash in ast_mutex_init() - (Patched by twilson) - - * Resolution of several DTMF based attended transfer issues. - (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, - shihchuan, grecco. Patched by rmudgett) - - NOTE: Be sure to read the ChangeLog for more information about these changes. - - * Resolve deadlocks related to device states in chan_sip - (Closes issue #18310. Reported, patched by one47. Patched by jpeeler) - - * Resolve an issue with the Asterisk manager interface leaking memory when - disabled. - (Reported internally by kmorgan. Patched by russellb) - - * Support greetingsfolder as documented in voicemail.conf.sample. - (Closes issue #17870. Reported by edhorton. Patched by seanbright) - - * Fix channel redirect out of MeetMe() and other issues with channel softhangup - (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. - Patched by russellb) - - * Fix voicemail sequencing for file based storage. - (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by - jpeeler) - - * Set hangup cause in local_hangup so the proper return code of 486 instead of - 503 when using Local channels when the far sides returns a busy. Also affects - CCSS in Asterisk 1.8+. - (Patched by twilson) - - * Fix issues with verbose messages not being output to the console. - (Closes issue #18580. Reported by pabelanger. Patched by qwell) - - * Fix Deadlock with attended transfer of SIP call - (Closes issue #18837. Reported, patched by alecdavis. Tested by - alecdavid, Irontec, ZX81, cmaj) - - Includes changes per AST-2011-005 and AST-2011-006 - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4 - - Information about the security releases are available at: - - http://downloads.asterisk.org/pub/security/AST-2011-005.pdf - http://downloads.asterisk.org/pub/security/AST-2011-006.pdf- The Asterisk Development Team has announced security releases for Asterisk - branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are - released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two - issues: - - * File Descriptor Resource Exhaustion (AST-2011-005) - * Asterisk Manager User Shell Access (AST-2011-006) - - The issues and resolutions are described in the AST-2011-005 and AST-2011-006 - security advisories. - - For more information about the details of these vulnerabilities, please read the - security advisories AST-2011-005 and AST-2011-006, which were 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-1.4.40.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.25 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.3 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.3 - - Security advisory AST-2011-005 and AST-2011-006 are available at: - - http://downloads.asterisk.org/pub/security/AST-2011-005.pdf - http://downloads.asterisk.org/pub/security/AST-2011-006.pdf- Bump release and rebuild for mysql 5.5.10 soname change.- The Asterisk Development Team has announced security releases for Asterisk - branches 1.6.1, 1.6.2, and 1.8. The available security releases are - released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - ** This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which - contained a bug which caused duplicate manager entries (issue #18987). - - The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues: - - * Resource exhaustion in Asterisk Manager Interface (AST-2011-003) - * Remote crash vulnerability in TCP/TLS server (AST-2011-004) - - The issues and resolutions are described in the AST-2011-003 and AST-2011-004 - security advisories. - - For more information about the details of these vulnerabilities, please read the - security advisories AST-2011-003 and AST-2011-004, which were 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-1.6.1.24 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.2 - - Security advisory AST-2011-003 and AST-2011-004 are available at: - - http://downloads.asterisk.org/pub/security/AST-2011-003.pdf - http://downloads.asterisk.org/pub/security/AST-2011-004.pdf- The Asterisk Development Team has announced security releases for Asterisk - branches 1.6.1, 1.6.2, and 1.8. The available security releases are - released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues: - - * Resource exhaustion in Asterisk Manager Interface (AST-2011-003) - * Remote crash vulnerability in TCP/TLS server (AST-2011-004) - - The issues and resolutions are described in the AST-2011-003 and AST-2011-004 - security advisories. - - For more information about the details of these vulnerabilities, please read the - security advisories AST-2011-003 and AST-2011-004, which were 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-1.6.1.23 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.1 - - Security advisory AST-2011-003 and AST-2011-004 are available at: - - http://downloads.asterisk.org/pub/security/AST-2011-003.pdf - http://downloads.asterisk.org/pub/security/AST-2011-004.pdf- The Asterisk Development Team has announced the release of Asterisk 1.8.3. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.3 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: - - * Resolve duplicated data in the AstDB when using DIALGROUP() - (Closes issue #18091. Reported by bunny. Patched by tilghman) - - * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. - (Closes issue #18464. Reported, patched by IgorG) - - * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of - unit tests for the function that does the parsing. - (Closes issue #18350. Reported by gbour. Patched by Marquis) - - * When using cdr_pgsql the billsec field was not populated correctly on - unanswered calls. - (Closes issue #18406. Reported by joscas. Patched by tilghman) - - * Resolve memory leak in iCalendar and Exchange calendaring modules. - (Closes issue #18521. Reported, patched by pitel. Tested by cervajs) - - * This version of Asterisk includes the new Compiler Flags option - BETTER_BACKTRACES which uses libbfd to search for better symbol information - within both the Asterisk binary, as well as loaded modules, to assist when - using inline backtraces to track down problems. - (Patched by tilghman) - - * Resolve issue where no Music On Hold may be triggered when using - res_timing_dahdi. - (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested - by francesco_r, rfrantik, one47) - - * Resolve a memory leak when the Asterisk Manager Interface is disabled. - (Reported internally by kmorgan. Patched by russellb) - - * Reimplemented fax session reservation to reverse the ABI breakage introduced - in r297486. - (Reported internally. Patched by mnicholson) - - * Fix regression that changed behavior of queues when ringing a queue member. - (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) - - * Resolve deadlock involving REFER. - (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.) - - Additionally, this release has the changes related to security bulletin - AST-2011-002 which can be found at - http://downloads.asterisk.org/pub/security/AST-2011-002.pdf - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3- - The Asterisk Development Team has announced the third release candidate of - Asterisk 1.8.3. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.3-rc3 resolves the following issues in addition to - those included in 1.8.3-rc1 and 1.8.3-rc2: - - * Fix regression that changed behavior of queues when ringing a queue member. - (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) - - * Resolve deadlock involving REFER. - (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc3- Bump release to build for F15- Remove isa macros- Make library dependencies architecture specific- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_RebuildThe Asterisk Development Team has announced the second release candidate of Asterisk 1.8.3. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.3-rc2 resolves the following issues in addition to those included in 1.8.3-rc1: * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) * Resolve a memory leak when the Asterisk Manager Interface is disabled. (Reported internally by kmorgan. Patched by russellb) * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported internally. Patched by mnicholson) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc2- - The Asterisk Development Team has announced the first release candidate of - Asterisk 1.8.3. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.3-rc1 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 candidate: - - * Resolve duplicated data in the AstDB when using DIALGROUP() - (Closes issue #18091. Reported by bunny. Patched by tilghman) - - * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. - (Closes issue #18464. Reported, patched by IgorG) - - * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of - unit tests for the function that does the parsing. - (Closes issue #18350. Reported by gbour. Patched by Marquis) - - * When using cdr_pgsql the billsec field was not populated correctly on - unanswered calls. - (Closes issue #18406. Reported by joscas. Patched by tilghman) - - * Resolve memory leak in iCalendar and Exchange calendaring modules. - (Closes issue #18521. Reported, patched by pitel. Tested by cervajs) - - * This version of Asterisk includes the new Compiler Flags option - BETTER_BACKTRACES which uses libbfd to search for better symbol information - within both the Asterisk binary, as well as loaded modules, to assist when - using inline backtraces to track down problems. - (Patched by tilghman)- - The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.2.3 resolves the following issue: - - * Reimplemented fax session reservation to reverse the ABI breakage introduced - in r297486. - (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by - mnicholson) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2.3- Build with SRTP support- - 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- - 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- - 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- - 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- - 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- dont package up the ices bits on el the client doesnt exist for us- dont build the 389 directory server package its not available on rhel6- dont always build AIS modules we dont have the BuildRequires on epel- Rebuild for new net-snmp.- Always build AIS modules- The Asterisk Development Team is proud to announce the release of Asterisk - 1.8.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - Asterisk 1.8 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.4. For more information about - support time lines for Asterisk releases, see the Asterisk versions page. - - http://www.asterisk.org/asterisk-versions - - The release of Asterisk 1.8.0 would not have been possible without the support - and contributions of the community. Since Asterisk 1.6.2, we've had over 500 - reporters, more than 300 testers and greater than 200 developers contributed to - this release. - - You can find a summary of the work involved with the 1.8.0 release in the - sumary: - - http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt - - A short list of available features includes: - - * Secure RTP - * IPv6 Support in the SIP channel driver - * Connected Party Identification Support - * Calendaring Integration - * A new call logging system, Channel Event Logging (CEL) - * Distributed Device State using Jabber/XMPP PubSub - * Call Completion Supplementary Services support - * Advice of Charge support - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup - - For a full list of changes in the current release candidate, please see the - ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0 - - Thank you for your continued support of Asterisk!- - The release of Asterisk 1.8.0-rc5 was triggered by some last minute platform - compatibility IPv6 changes. In addition, the availability of the English sound - prompts with Australian accents has been added. - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup - - For a full list of changes in the current release candidate, please see the - ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5 - - This release candidate contains fixes since the last release candidate as - reported by the community. A sampling of the changes in this release candidate - include: - - * Additional fixups in chan_gtalk that allow outbound calls to both Google - Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip - and stunaddr. - (Closes issue #13971. Patched by dvossel) - - * Resolve manager crash issue. - (Closes issue #17994. Reported by vrban. Patchd by dvossel) - - * Documentation updates for sample configuration files. - (Closes issues #18107, #18101. Reported, patched by lathama, lmadsen) - - * Resolve issue where faxdetect would only detect the first fax call in - chan_dahdi. - (Closes issue #18116. Reported by seandarcy. Patched by rmudgett) - - * Resolve issue where a channel that is setup and torn down *very* quickly may - not have the right call disposition or ${DIALSTATUS}. - (Closes issue #16946. Reported by davidw. Review - https://reviewboard.asterisk.org/r/740/) - - * Set TCLASS field of IPv6 header when SIP QoS options are set. - (Closes issue #18099. Reported by jamesnet. Patched by dvossel) - - * Resolve issue where Asterisk could crash on shutdown when using SRTP. - (Closes issue #18085. Reported by st. Patched by twilson) - - * Fix issue where peers host port would be lost on a SIP reload. - (Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel) - - A short list of available features includes: - - * Secure RTP - * IPv6 Support in the SIP channel driver - * Connected Party Identification Support - * Calendaring Integration - * A new call logging system, Channel Event Logging (CEL) - * Distributed Device State using Jabber/XMPP PubSub - * Call Completion Supplementary Services support - * Advice of Charge support - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup - - For a full list of changes in the current release candidate, please see the - ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4- This release candidate contains fixes since the release candidate as reported by - the community. A sampling of the changes in this release candidate include: - - * Still build chan_sip even if res_crypto cannot be built (use, but not depend) - (Reported by a user on the mailing list. Patched by tilghman) - - * Get notifications for call files only when a file is closed, not when created - (Closes issue #17924. Reported by mkeuter. Patched by abeldeck) - - * Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk - expects the DTMF to arrive on the RTP stream and not via jingle DTMF - signalling. - (Patched by dvossel. Tested by malcolmd) - - * Fixes to allow chan_gtalk to communicate with the Gmail web client. - (Patched by phsultan and dvossel) - - * Fix to GET DATA to allow audio to be streamed via an AGI. - (Closes issue #18001. Reported by jamicque. Patched by tilghman) - - * Resolve dnsmgr memory corruption in chan_iax2. - (Closes issue #17902. Reported by afried. Patched by russell, dvossel) - - A short list of available features includes: - - * Secure RTP - * IPv6 Support in the SIP channel driver - * Connected Party Identification Support - * Calendaring Integration - * A new call logging system, Channel Event Logging (CEL) - * Distributed Device State using Jabber/XMPP PubSub - * Call Completion Supplementary Services support - * Advice of Charge support - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout - - For a full list of changes in the current release candidate, please see the - ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc3- This release candidate contains fixes since the last beta release as reported by - the community. A sampling of the changes in this release candidate include: - - * Add slin16 support for format_wav (new wav16 file extension) - (Closes issue #15029. Reported, patched by andrew. Tested by Qwell) - - * Fixes a bug in manager.c where the default configuration values weren't reset - when the manager configuration was reloaded. - (Closes issue #17917. Reported by lmadsen. Patched by bbryant) - - * Various fixes for the calendar modules. - (Patched by Jan Kalab. - Reviewboard: https://reviewboard.asterisk.org/r/880/ - Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/ - Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/) - - * Add CHANNEL(checkhangup) to check whether a channel is in the process of - being hung up. - (Closes issue #17652. Reported, patched by kobaz) - - * Fix a bug with MeetMe where after announcing the amount of time left in a - conference, if music on hold was playing, it doesn't restart. - (Closes issue #17408, Reported, patched by sysreq) - - * Fix interoperability problems with session timer behavior in Asterisk. - (Closes issue #17005. Reported by alexcarey. Patched by dvossel) - - * Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was - determined to be one of the most significant bottlenecks in SIP registration - processing. This patch improved the speed of an astdb load test by 50000% - (yes, Fifty-Thousand Percent). On this particular load test setup, this - doubled the number of SIP registrations the server could handle. - (Review: https://reviewboard.asterisk.org/r/825/) - - * Don't clear the username from a realtime database when a registration - expires. Non-realtime chan_sip does not clear the username from memory when a - registration expiries so realtime probably shouldn't either. - (Closes issue #17551. Reported, patched by: ricardolandim. Patched by - mnicholson) - - * Don't hang up a call on an SRTP unprotect failure. Also make it more obvious - when there is an issue en/decrypting. - (Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by - twilson) - - * Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5! - - A short list of available features includes: - - * Secure RTP - * IPv6 Support in the SIP channel driver - * Connected Party Identification Support - * Calendaring Integration - * A new call logging system, Channel Event Logging (CEL) - * Distributed Device State using Jabber/XMPP PubSub - * Call Completion Supplementary Services support - * Advice of Charge support - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout - - For a full list of changes in the current release candidate, please see the - ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc2- This release contains fixes since the last beta release as reported by the - community. A sampling of the changes in this release include: - - * Fix issue where TOS is no longer set on RTP packets. - (Closes issue #17890. Reported, patched by elguero) - - * Change pedantic default value in chan_sip from 'no' to 'yes' - - * Asterisk now dynamically builds the "Supported" header depending on what is - enabled/disabled in sip.conf. Session timers used to always be advertised as - being supported even when they were disabled in the configuration. - (Related to issue #17005. Patched by dvossel) - - * Convert MOH to use generic timers. - (Closes issue #17726. Reported by lmadsen. Patched by tilghman) - - * Fix SRTP for changing SSRC and multiple a=crypto SDP lines. Adding code to - Asterisk that changed the SSRC during bridges and masquerades broke SRTP - functionality. Also broken was handling the situation where an incoming - INVITE had more than one crypto offer. - (Closes issue #17563. Reported by Alexcr. Patched by twilson) - - Asterisk 1.8 contains many new features over previous releases of Asterisk. - A short list of included features includes: - - * Secure RTP - * IPv6 Support in the SIP Channel - * Connected Party Identification Support - * Calendaring Integration - * A new call logging system, Channel Event Logging (CEL) - * Distributed Device State using Jabber/XMPP PubSub - * Call Completion Supplementary Services support - * Advice of Charge support - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta5- This release contains fixes since the last beta release as reported by the - community. A sampling of the changes in this release include: - - * Fix parsing of IPv6 address literals in outboundproxy - (Closes issue #17757. Reported by oej. Patched by sperreault) - - * Change the default value for alwaysauthreject in sip.conf to "yes". - (Closes issue #17756. Reported by oej) - - * Remove current STUN support from chan_sip.c. This change removes the current - broken/useless STUN support from chan_sip. - (Closes issue #17622. Reported by philipp2. - Review: https://reviewboard.asterisk.org/r/855/) - - * PRI CCSS may use a stale dial string for the recall dial string. If an - outgoing call negotiates a different B channel than initially requested, the - saved original dial string was not transferred to the new B channel. CCSS - uses that dial string to generate the recall dial string. - (Patched by rmudgett) - - * Split _all_ arguments before parsing them. This fixes multicast RTP paging - using linksys mode. - (Patched by russellb) - - * Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure - data has valid CSV formatting. Also list the special CEL variables that are - available for use in the mapping. There are also several other CEL fixes in - this release. - (Patched by russellb) - - Asterisk 1.8 contains many new features over previous releases of Asterisk. - A short list of included features includes: - - * Secure RTP - * IPv6 Support in the SIP Channel - * Connected Party Identification Support - * Calendaring Integration - * A new call logging system, Channel Event Logging (CEL) - * Distributed Device State using Jabber/XMPP PubSub - * Call Completion Supplementary Services support - * Advice of Charge support - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta4- - This release contains fixes since the last beta release as reported by the - community. A sampling of the changes in this release include: - - * Fix a regression where HTTP would always be enabled regardless of setting. - (Closes issue #17708. Reported, patched by pabelanger) - - * ACL errors displayed on screen when using dynamic_exclude_static in sip.conf - (Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson) - - * Support "channels" in addition to "channel" in chan_dahdi.conf. - (https://reviewboard.asterisk.org/r/804) - - * Fix parsing error in sip_sipredirect(). The code was written in a way that - did a bad job of parsing the port out of a URI. Specifically, it would do - badly when dealing with an IPv6 address. - (Closes issue #17661. Reported by oej. Patched by mmichelson) - - * Fix inband DTMF detection on outgoing ISDN calls. - (Patched by russellb and rmudgett) - - * Fixes issue with translator frame not getting freed. This issue prevented - g729 licenses from being freed up. - (Closes issue #17630. Reported by manvirr. Patched by dvossel) - - * Fixed IPv6-related SIP parsing bugs and updated documention. - (Reported by oej. Patched by sperreault) - - * Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a - list of a specified item. Matches up with FIELDQTY() and CUT(). - (Closes #17713. Reported, patched by gareth. Tested by tilghman) - - Asterisk 1.8 contains many new features over previous releases of Asterisk. - A short list of included features includes: - - * Secure RTP - * IPv6 Support - * Connected Party Identification Support - * Calendaring Integration - * A new call logging system, Channel Event Logging (CEL) - * Distributed Device State using Jabber/XMPP PubSub - * Call Completion Supplementary Services support - * Advice of Charge support - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta3- Rebuild against libpri 1.4.12- Update to 1.8.0-beta2 - Disable building chan_misdn until compilation errors are figured out (https://issues.asterisk.org/view.php?id=14333) - Start stripping tarballs again because Digium added MP3 code :(- - The following are a few of the issues resolved by community developers: - - * Allow users to specify a port for DUNDI peers. - (Closes issue #17056. Reported, patched by klaus3000) - - * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is - set. - (Closes issue #16815. Reported, patched by rain) - - * If there is realtime configuration, it does not get re-read on reload unless - the config file also changes. - (Closes issue #16982. Reported, patched by dmitri) - - * Send AgentComplete manager event for attended transfers. - (Closes issue #16819. Reported, patched by elbriga) - - * Correct manager variable 'EventList' case. - (Closes issue #17520. Reported, patched by kobaz) - - In addition, changes to res_timing_pthread that should make it more stable have - also been implemented. - - For a full list of changes in the current release, please see the - ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10- Add patch to remove requirement on latex2html- Mass rebuild with perl-5.12.0- * Fix building CDR and CEL SQLite3 modules. - (Closes issue #17017. Reported by alephlg. Patched by seanbright) - - * Resolve crash in SLAtrunk when the specified trunk doesn't exist. - (Reported in #asterisk-dev by philipp64. Patched by seanbright) - - * Include an extra newline after "Aliased CLI command" to get back the prompt. - (Issue #16978. Reported by jw-asterisk. Tested, patched by seanbright) - - * Prevent segfault if bad magic number is encountered. - (Issue #17037. Reported, patched by alecdavis) - - * Update code to reflect that handle_speechset has 4 arguments. - (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger, - mmichelson) - - * Resolve a deadlock in chan_local. - (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2)- Update to 1.6.2.7-rc3- Update to 1.6.2.7-rc2- Update to final 1.6.2.6 - - The following are a few of the issues resolved by community developers: - - * Make sure to clear red alarm after polarity reversal. - (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, - Chainsaw, mikeeccleston) - - * Fix problem with duplicate TXREQ packets in chan_iax2 - (Closes issue #16904. Reported, patched by rain. Tested by rain, dvossel) - - * Fix crash in app_voicemail related to message counting. - (Closes issue #16921. Reported, tested by whardier. Patched by seanbright) - - * Overlap receiving: Automatically send CALL PROCEEDING when dialplan starts - (Reported, Patched, and Tested by alecdavis) - - * For T.38 reINVITEs treat a 606 the same as a 488. - (Closes issue #16792. Reported, patched by vrban) - - * Fix ConfBridge crash when no timing module is loaded. - (Closes issue #16471. Reported, tested by kjotte. Patched, tested by junky) - - For a full list of changes in this releases, please see the ChangeLog: - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.6- Update to 1.6.2.6-rc2- Add a patch that fixes CLI history when linking against external libedit.- Update to 1.6.2.5 - - * AST-2010-002: Invalid parsing of ACL rules can compromise security- Update to 1.6.2.4 - - * AST-2010-002: This security release is intended to raise awareness - of how it is possible to insert malicious strings into dialplans, - and to advise developers to read the best practices documents so - that they may easily avoid these dangers.- Update to 1.6.2.2 - - * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can - remotely crash Asterisk by modifying the FaxMaxDatagram field of - the SDP to contain either a negative or exceptionally large value. - The same crash occurs when the FaxMaxDatagram field is omitted from - the SDP as well.- Update to 1.6.2.1 final: - - * CLI 'queue show' formatting fix. - (Closes issue #16078. Reported by RoadKill. Tested by dvossel. Patched by - ppyy.) - - * Fix misreverting from 177158. - (Closes issue #15725. Reported, Tested by shanermn. Patched by dimas.) - - * Fixes subscriptions being lost after 'module reload'. - (Closes issue #16093. Reported by jlaroff. Patched by dvossel.) - - * app_queue segfaults if realtime field uniqueid is NULL - (Closes issue #16385. Reported, Tested, Patched by haakon.) - - * Fix to Monitor which previously assumed the file to write to did not contain - pathing. - (Closes issue #16377, #16376. Reported by bcnit. Patched by dant.- Update to 1.6.2.1-rc1- Released version of 1.6.2.0- Update to 1.6.2.0-rc8- Update to 1.6.2.0-rc7- Change the logrotate and the init scripts so that Asterisk doesn't try and write to / or /root- Make dependency on uw-imap conditional and some other changes to make building on RHEL5 easier.- Update to 1.6.2.0-rc6- Update to 1.6.2.0-rc5- Update to 1.6.2.0-rc4- Add patch from upstream to fix how res_http_post forms paths.- Add an AST_EXTRA_ARGS option to the init script - have the init script to cd to /var/spool/asterisk to prevent annoying message- Compile against gmime 2.2 instead of gmime 2.4 because the patch to convert the API calls from 2.2 to 2.4 caused crashes.- Require latex2html used in static-http documents- Change ownership and permissions on config files to protect them.- Update to 1.6.2.0-rc3- Merge firmware subpackage back into the main package.- Package internal help. - Fix up some more paths in the configs so that everything ends up where we want them.- Update to 1.6.2.0-rc2 - We no longer need to strip the tarball as it no longer includes non-free items.- Enable building of API docs. - Depend on version 1.2 or newer of speex- Update to 1.6.1.6 - Drop patches that are too troublesome to maintain anymore or have been integrated upstream.- Add a patch from Quentin Armitage and rebuld.- rebuilt with new openssl- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild- Rebuild to pick up new AIS and ODBC deps. - Update script that strips out bad content from tarball to do the download and to check the GPG signature.- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild- Update to 1.6.1-rc1 - Add backport of conference bridging that is slated for 1.6.2 - Add patches to conference bridging that implement CLI apps- rebuild with new openssl- Fedora Directory Server compatibility patch/subpackage.- Fix up paths. BZ#477238- Update patches- Update to 1.6.1-beta4- Update to 1.6.1-beta3- Rebuild for new gmime- Add patch to fix missing variable on PPC.- Update PPC systems don't have sys/io.h patch.- PPC systems don't have sys/io.h- Update to 1.6.1 beta 2- Fix issue with init script giving wrong path to config file.- Explicitly require dahdi-tools-libs to see if we can get this to build.- Update to final release.- Rebuild- Replace app_rxfax/app_txfax with app_fax taken from upstream SVN.- Bump release and rebuild with new libpri and zaptel.- Add patch pulled from upstream SVN that fixes AST-2008-010 and AST-2008-011.- Add patch for LDAP extracted from upstream SVN (#442011)- Add patch that unbreaks cdr_tds with FreeTDS 0.82. - Properly obsolete conference subpackage.- Disable building cdr_tds since new FreeTDS in rawhide no longer provides needed library.- Bump release and rebuild to fix libtds breakage.- Update to 1.6.0-beta9. - Update patches so that they apply cleanly. - Temporarily disable app_conference patch as it doesn't compile - config/scripts/postgres_cdr.sql has been merged into realtime_pgsql.sql - Re-add the asterisk-strip.sh script as a source file.- Update to 1.6.0-beta8 - Contains fixes for AST-2008-006 / CVE-2008-1897- Return to stripped tarballs since there's more non-free content in the Asterisk tarballs than I thought.- Update to 1.6.0-beta7.1 - Update patches - Back out some changes that were made because beta7 was tagged from the wrong branch.- Update to 1.6.0-beta7 - The Asterisk tarball no longer contains the iLBC code, so we can directly use the upstream tarball without having to modify it. - Get rid of the asterisk-strip.sh script since it's no longer needed. - Diable build of codec_ilbc and format_ilbc (these do not contain any legally suspect code so they can be included in the tarball but it's pointless building them). - Update chan_mobile patch to fix API breakages. - Add a patch to chan_usbradio to fix API breakages.- Add Postgresql schemas from contrib as documentation to the Postgresql subpackage.- Update patches. - Add patch to compile against external libedit rather than using the in-tree version. - Add -Werror-implicit-function-declaration to optflags. - Get rid of hashtest and hashtest2 binaries that link to unfortified versions of *printf functions. They are compiled with -O0 which somehow pulls in the wrong versions. These programs aren't necessary to the operation of the package anyway.- Update to 1.6.0-beta6 to fix some security issues. - - AST-2008-002 details two buffer overflows that were discovered in - RTP codec payload type handling. - * http://downloads.digium.com/pub/security/AST-2008-002.pdf - * All users of SIP in Asterisk 1.4 and 1.6 are affected. - - AST-2008-003 details a vulnerability which allows an attacker to - bypass SIP authentication and to make a call into the context - specified in the general section of sip.conf. - * http://downloads.digium.com/pub/security/AST-2008-003.pdf - * All users of SIP in Asterisk 1.0, 1.2, 1.4, or 1.6 are affected. - - AST-2008-004 Logging messages displayed using the ast_verbose - logging API call are not displayed as a character string, they are - displayed as a format string. - * http://downloads.digium.com/pub/security/AST-2008-004.pdf - - AST-2008-005 details a problem in the way manager IDs are caculated. - * http://downloads.digium.com/pub/security/AST-2008-005.pdf- add Requires for versioned perl (libperl.so)- Update to 1.6.0-beta5 - Remove upstreamed patches.- Package the directory used to store monitor recordings.- Add patch from David Woodhouse that fixes building on PPC64.- Update to 1.6.0 beta 4- Update to 1.4.18. - Use -march=i486 on i386 builds for atomic operations (GCC 4.3 compatibility). - Use "logger reload" instead of "logger rotate" in logrotate file (#432197). - Don't explicitly specify a group in in the init script to prevent Zaptel breakage (#426629). - Split app_ices out to a separate package so that the ices package can be required. - pbx_kdeconsole has been dropped, don't specifically exclude it from the build anymore. - Update app_conference patch. - Drop upstreamed libcap patch.- Update to 1.4.17 to fix AST-2008-001.- Update to 1.4.16.2- Bump release and rebuild to fix broken dep on uw-imap.- Update to the real 1.4.16.1.- Add patch to bring source up to version 1.4.16.1 which will be released shortly to fix some crasher bugs introduced by 1.4.16.- Update to 1.4.16 to fix security bug.- Really, really fix the build problems on devel.- Tweaks to get to build on x86_64- Exclude PPC64- Don't build apidocs by default since there's a problem building on x86_64.- Really get rid of zero length map files.- Get rid of zero length map files. - Shorten descriptions of voicemail subpackages- Update to 1.4.15- Fix license and other rpmlint warnings.- Update to 1.4.14- Add chan_mobile- Don't build cdr_sqlite because sqlite2 has been orphaned. - Rebase local patches to latest upstream SVN - Update app_conference patch to latest from upstream SVN - Apply post-1.4.13 patches from upstream SVN- Update to 1.4.13- Update to 1.4.12.1- Update to 1.4.11- Update to 1.4.10.1.- Update to 1.4.10 (security update).- Add a patch that allows alternate extensions to be defined in users.conf- Update app_conference patch. Enter/leave sounds are now possible.- Update patches so we don't need to run auto* tools, because autoconf 2.60 is required and FC-6 and RHEL5 only have autoconf 2.59.- Don't build app_mp3- Add app_conference- Use plain useradd/groupadd rather than the fedora-usermgmt - Clean up requirements - Clean up build requirements by moving them to package sections- Update to 1.4.9- Update to 1.4.8 - Drop ixjuser patch.- Update to 1.4.7.1- Update to 1.4.7 - RxFAX/TxFAX applications- It's "sbin", not "bin" silly.- Add patch that lets us change TOS bits even when running non-root- voicemail needs to require /usr/bin/sox and /usr/bin/sendmail- Update to 1.4.6 - Remove upstreamed patch.- Build the IMAP and ODBC storage options of voicemail and split voicemail out into subpackages. - Apply patch so that the system UW IMAP libray can be linked against. - Patch modules.conf.sample so that alternal voicemail modules don't get loaded simultaneously. - Link against system GSM library rather than internal copy. - Patch the Makefile so that it doesn't add redundant/wrong compiler options. - Force building with the standard RPM optimization flags. - Install the Asterisk MIB in a location that net-snmp can find it. - Only package docs in the main package that are relevant and that haven't been packaged by a subpackage. - Other minor cleanups.- Move sounds- Update some more ownership/permissions- Fix some permissions.- Update init script patch - Move pid file to subdir of /var/run- Update init script patch to run as non-root- Build modules that depend on FreeTDS. - Don't build voicemail with ODBC storage.- Have the build output the commands executing, rather than covering them up.- Update to 1.4.5 - Remove upstreamed patch.- Add a patch to fix CVE-2007-2488/ASA-2007-013- Update to 1.4.4- Update to 1.4.2- Package the IAXy firmware - Minor clean-ups in files- Update to 1.4.1 - Don't build/package codec_zap (zaptel 1.4.0 doesn't support it)- Update to 1.4.0-beta4 - Various cleanups.- Don't package IAXy firmware because of license - Don't build app_rpt - Don't BR lm_sensors on PPC - Better way to prevent download/installation of sound archives - Redo tarball to eliminate non-free items- Remove explicit dependency on glibc-kernheaders. - Build jabber modules on PPC- *Really* update to beta3 - chan_jingle has been taken out of 1.4 - Move misplaced binaries to where they should be- Remove requirement on asterisk-sounds-core until licensing can be figured out.- Update to 1.4.0-beta3- Update to 1.4.0-beta2- Update to 1.2.10.- Update to 1.2.9.1- Update to 1.2.8 - Add misdn.conf to list of configs. - Drop chan_bluetooth patch for now...- Zaptel subpackage shouldn't obsolete the sqlite subpackage. - Remove mISDN until build issues can be figured out.- Build mISDN channel drivers, modelled after spec file from David Woodhouse- Update chan_bluetooth patch with some additional information as to it's source and comment out more in the configuration file.- Add chan_bluetooth- Split off more stuff into subpackages.- Update to 1.2.7- Fix detection of libpri on 64 bit arches (taken from Matthias Saou's rpmforge package) - Change sqlite subpackage name to sqlite2 (there are sqlite3 modules in development).- Don't build GTK 1.X console since GTK 1.X is being moved out of core...- Update to 1.2.6- Update to 1.2.5. - Removed upstreamed MOH patch. - Add full urls to the app_(r|t)xfax.c sources. - Update spandsp patch.- Actually apply the patch.- Add patch to keep Asterisk from crashing when using MOH inside a MeetMe conference.- BR sqlite2-devel- Update to 1.2.4.- Took some tricks from Asterisk packages by Roy-Magne Mo. - Enable gtk console module. - BR gtk+-devel. - Add logrotate script. - BR sqlite2-devel and new sqlite subpackage. - BR doxygen and graphviz for building duxygen documentation. (But don't build it yet.)- Completely eliminate the "asterisk" user from the spec file. - Move more config files to subpackages. - Consolidate two patches that patch the init script. - BR curl-devel - BR alsa-lib-devel - alsa, curl, oss subpackages- Do not run as user "asterisk" as that prevents setting of IP TOS (which is bad for quality of service). - Add patch for setting TOS separately for SIP and RTP packets.- First version for Fedora Extras.18.12.1-1.el8.218.12.1-1.el8.218.12.1-1.el8.2.build-iddf14b9e37ef5d09ffa5e81f55f40c7c0c56ed0.2dcdcc1d9197f21318f38ce16e51a2397f03be3.2app_directory_plain.soapp_voicemail_plain.so/usr/lib//usr/lib/.build-id/7d//usr/lib/.build-id/a3//usr/lib64/asterisk/modules/-O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -Werror-implicit-function-declaration -DLUA_COMPAT_MODULE -fPICcpioxz2x86_64-redhat-linux-gnudirectorysymbolic link to ../../../../usr/lib64/asterisk/modules/app_directory_plain.soELF 64-bit LSB shared object, x86-64, version 1 (SYSV), dynamically linked, BuildID[sha1]=a3dcdcc1d9197f21318f38ce16e51a2397f03be3, strippedELF 64-bit LSB shared object, x86-64, version 1 (SYSV), dynamically linked, BuildID[sha1]=7ddf14b9e37ef5d09ffa5e81f55f40c7c0c56ed0, stripped R RRRRRRRR R RRRRRRRR https://bugz.fedoraproject.org/asteriskutf-8a0beaaaecd55038496137e7ae58c304e0e1268a641cd752b303cadbe380c8787?7zXZ !#,}] b2u jӫ`(y.r#rlрRߏa˰-|Ţ%ac-qN+v)غx~ٔG#M-CGPhڛioo,B{ HStqtQR"9?R@,j&ˑ/JéLm|)kٱP\8(hPF2W!A*ϔ\NdKFh)Y 5okGޙ^Q'nΈRԝݗk;3iBQ/4' ~Ba~;2!gPEtCK8(:b갌5[!PX76enM̒s1Vϊͅ|7a)"ESȪ^G ,qyc:5u 5\tKڭ OURe23 Kt(3 7t۶eϋḙCȾ AG T>xx0hhr&kB+ x8[d=b?`>~.w^ .MyliM)zim"^XEs^~nAcFHj+FB)UٛN7+&oqh}&S%:k[jI7 }&4] r;*AyH<:N` \[:dTiMy9yOIxkyn^5elg`RjRErBsx)y!P7bgc| A C JٯOa$E!k\ֹM{ /c48Z{%2X=,:J\= 7qkOi/VF;"מU%Z9 OzУ٩"-VUvg>T@ ,Ғ'RA=#0ˣDQT#9`?b}/p^x'N\˯z`oUܧ- !dcj0!QƊL[ˈI`\y 2Ynn(:ǿtPnh { h+E۩ D1$WYc*KWNBEL |gЈeUL"LoB DHOwܔ-(u\=WpGlKġHʙ꼂~ [3?е?=/T;Ћ AA)olQznjfnOzB$,9~gYضޠMJxŤ ~D忺#v@R_b0E 4*&?mTҮyd~D>/+h {Z(S/VR&^x0}N׃Ԫ:A.]qP.O@L\64`"Ӡ,9zJ9=N> >ºN[N]cHx["g3o"L[ jXM#f/Q ;Y_3-H;Z ~;G?9ptd=߿>"f^嵽02dEFPIy0dZI#5dO/,1vqAO7 E}Z8ߐzYox=FA\#EMx>ug"r Pk%hw2k>QN<^ d7z]m2[Sm@OC )6[-B-ⶦ:,*3̙?|3 E;o3` 6{0~2{JZa@8lgIat[,/g*ZAN(2ۊ7}X,X'n(}U_Ł>5m'/t\K$)Y}zv󗹾R7Ԭ b/e~τ+R6t8j `9b M dY "%nDw87%d}7cFbd^h)e"x*tDUЏ6r._!-&/LDS3}oJhR]xSbJC\ܛ=t>oĢ*8oxma[l*KTírY:! H!V%4;>$'}h:-N 2Y*5*zp 9ס'XTUg}TMR3h4ȑP8} ̬AV擉.[瓄r#\Ab h'tlt Y$idYL}ՁbY!Cwz/Vڜt:~!ߎ-Ƃ^2 ={!_ò\ nDlAn\!V9=N+v^nYB]X2>7_1rToy4Ia3)!>z&bUP& O;<_(!=9rɣr:(j[:PYM2 [HT"@| iM_W":vޠ/=ocq%pNt0KC&W*CmSU=?,A4B-8`u ~H0wFT A<o!x1Db74rrq qVG'W:ϓڸv <֍X/pP]II}5]+5CU=/RYpχGvob&q1f#^皈Hf/+- #>e%V8 r"P^'*5ܴ/R{FI#o5\ BO׵N2Ou&iw QGGRALm:Lp<1I|[cnT2;V "R&b΅ħɭX"5盈Oi6[لz U||>9v$=Y |D|~pEvF` QCJ["uzqVR;Pk-l1,< e>/ˁ(-H|aTOY,wThl7dK8 ݅Kk$ &|qF៎AAxduEk1Tg"=PBuUH?Zv1*OP#Jhu2l& ɷPK L[n stF-MKK֕i5S OچTsQV ?!"1펇!4ҬG]x*kyI ֩K9Kv_=>dH^g嬶DUI(OFj}pg,9;Z\%Jqb_yM*WGH4GP,<{uL>JcUHj&POwBh[pL7rak93( QQ81 xpbpB@}PV7~:"s](O4um\CrCjna.6py 0H]} }R+*O*!{5d9ۄS; GmZ>/;~(<. $G(^<y,C3:qgD [,r$ 嵲i7Mʷ$c llSaImw66҅R_ȊD^;+‡9)w, X ½D9K ):R1̱(|\a4䝐̥tooT5WVQ@T:@./@pQ޵Xoe\ï@%vtYP ཫ Êd~deU$ +3bfm0 rLVFWHm\\߀ gUt@vę=&2DS|ENWx A]E Xcz ١!Be'+dFC6_1!q[# \{"ȲB:2.mqޘpCY3VGqbO)r}l{)Jڛ.F~^7^+ĒKae㛥|xtĄzTL5AuSzڠS_kx" t#* v q3o<'x \ qgvչ}̽Z@l5akan9Hƙ-+d]3 y0@c+"#<,om;lɮwc lZ |-fI| 6l`&B~)Uva#G[@E3(IVn.>jD힎KEW+भbQ@V/-2'bvsN/$'O ڊ`ܯ]@wjuqK/ɀ{Q@K`F~X*BN+d>]*RY 53K2cCLgoi)12*̬ s{ ̇~{Oa8`xyʟD;('qn q Krotw_#3[ ٠eBpTYZON7̠1q ;l[kO}yN:OYV{awsЗIZ,`1(7}52xDp"SRsQBiHS饄}Ik((NZJfZ=A**V ˶pyao'FzM;G @}t[xV@ˬ3I>7/@SaaZ蕡laDᘰ.)LosT D ɯaWIRHu_FצJٍ^&x]Dž >ۿ^(VAލgh^ː⠺.eBLR[:ß.YTdsb#`iNU>h@/!kbϔw8y*cX|Eϼp^>2:§Wo+v{OiʕmΓ~7LzyF*$1 4ŅMA,@>Mh$~4x m|hc{W`LpTx 魾Y&qOjNBO;y/8(t ЋbBZ犥'>-K0椯pl/A~*a a7_|b"\f)^dR,`4Sy/lL2FG/>  J95XĈe{"V(*_Viv᷍I$~0b𗇢HL+cڵmX/Lrq_r`9qLGze,Łc;yd5&A}S5\OŇJKv`swv1]a)y?#i{9aLZ.%ݫZ}0úV5R"/>\ h{~ZdxOњDY%L򸖡)~u<޴u:~؅B[6Q\aⲃAGCoOR:6-rFJy2WWB@TGM2|? U`׬e_CJāQ95 bGBD ŵ'&Uf ѢCῖAi DԿ\-^FE0YِsԟDr1?C ,5i1'mY[!rD{,:6S?-D!RV<}z9lьjZk],`={ Rm%-2}[zZ*VjV.:!Ѽ >φf\:Qu5PА5s+4/dX i c0=}Hz mzR\ҭ Vs櫺6^6OPOd%rKT:QpT0O,YVhEOii ۹6)('n;ʓ2-U3E)fm?c-tkd 95.RU4k05RNd3Fc>#sYK8M߹25pxI_"}Qhr\s8im)qT$wocX6S1,e9cmYb`#!9~*|3HAp* Oe^a䈣iwx'`TuZ.vm?n=.'I_J0"$5pp6DK,-;vYY5?{kKT'<%U:b2Vs/+ArB?>v] }tK)0RY٬B+q0|vجu4vs&.: OL#K,g-\حgIЦS[5\شI!~i{Lsp5ߘ9-: G]g%!.WQ;k6taȵsuқީ t4 { JoLKqqtW%|#F@N]5vL# ZhMK;P,?:QUffT)\V7}@:%\#n[UvȊ%' dӯ} Ӽ)1.Ύ% ԧ8@>?㪈4. 64AzԪ!i7Gx# Fͺk!nLMX ^GKlJ^w:SecS*R47owW-īVશB㯄tז@6&Ɉ$w 1Z1 U:{$ăʒ9*TjϬ8^`^1;p]Ӆ\sGΞ]Qp/X{R.΁ֽQ` hta,ip.ɼms}!6@NAZ .jd8ퟅWq_$)'\YU5ii2_8rH@Ɨ;}1%TWϴT>%8:*$ ! 08ON&~hu"yV⻁M(n$]q67}`EA9Ԛ=&FʪSaDFbo` g+oopY/ (o\#9H🤉U5ԭ0,Z.`3ayh9:Xq)n#@n7dr#|+w5h)|B͊XA_r_nf tCdBW/j*Y8`BBdKg&.ACztu?}W Fw.C!IJ%۽<&q\lj=q&cW[ML槹7l^)N"eF")_~ փt<%mG._$nׄ?D>42_"]{^%3]* l_~c{VR։FVbm&20b]6rvL_b>CyDz)3 Uzv&+l}Tg۴g^52K_RhEN(I1H3|{|DK,J4d 9wwٷ`uaͬsC^T?fۓ9FL+y!ؐ\#pkQsc@ƞːĮ* Z.0F`pi;Ƃհ&W$]um60`~Hurؽse.\ݷ'0dw1O+6nˊŽL(E8O)pΗFh L@O6fc+&VCca(ckVks#`E lMK!@q%dՙ#}>l\Xw&" cxt*yUz+˕Â01 x3xVr Gʪ²I%u3 Hr~ZA%*X$A2.mMVԌ8(74F^7VS M rj" R(q:hR5dla >ӑـ  H'G?Eley&ΌMfZVGK3^nVSÀ![XwU-^\:bW)l|H#Tr5WQ!"34%c߈qz%K P=esc4 hyQ!,Z|c F ML%b(Ӊ.RgUL#6; zX`폘xԦM$,OBPG3Nr\csʣ=+%0H̓SKAE}?=); tJ5|,0VP% DY)֗) DZ%=+b;pKNZ9݌KVp#N׬=iSjߣ;e/w톊 CnmS]D)&&vfΑKT6*6\`{qi=6IK4Y4+^u0dBMILS9X+ b=__5C VhF]V825acւfjH*_BXTMLI b>YWeKI?X 0 Z5ƕM:X/P,K] 7-zybzw O4`Ȍ|S*]_k۹l(x,)h8W[=ؼls ih0^W%,֜TvE*ӧvmsP(+Zr[1ĉ^ 㔙䘓U GB[@җ@CNe{pkm0(m/,8b^#oZRUad|>ӻbÕiA.;Vsozڣ(%]M\1> V d6 @/je1YͶ f/ ^dcu&; $n"o*sB~S0;u{o&]  pO;ia&@&9I}b2 UntE2$Jp]x~*c8ܸ*I0DTGz#$5VHș>4][N!r,%:uDWס/E\$q8k|y`F9V6%X]9yYU/F_YĬ[.ZQm D^(Ez]۪9S| &ީ܄f/-^w]'1vƐpẹ[JEIM&E]^:t^tkjSzNIqII@*Zf77͕2^Fw&/N.hIo-f-Wdq_Ƒ 3Qы;K HVka^4f/ K7f_o,(,TAϛw{4՗7~K ^ۂFv//doN4N$;[nݹW4>tLynH;j7цsJ]uC_>EWU]louS"Ei1[ߐq`TU0I$~ jK6쭶[ŵ6ZP㌼Tfz}uK|PÚ}RrI>WO9TgF,[}txZ @7l g-R2A`#\MRPqab#_3# }l![,{p:MPbD"׻'B<,OͽbLRdYilf -$auF#nE +^q_E}WEmy]0W36qkNA EPnTiO zA&&$hmfV]a)EPC\-Gf-is_f{0 f']Tݩ;)-k lBDOl5YKNHk;0wg9eb>eɂI>jG\lp)L=Lnm&[žB-sq/uxs I0eXSȘ_Ӆ.7O ȱePK`#晬Fހ]ZR4DhV1~HYJ,g+2UGTo]j9h ) XVx$愛&QGfz`.-٢v > ʅ1 u=2 qdȒ̰&z|y/J8&hO t%.Å#cO`P_<_Aۖ2jWg`g %7(k/<{ leP]=E2쾤pT…!AZ绨!b8G,a uGHҨni>5;I)/N| \y|^_ff쭥WAN-]mz>hhzdjas?|$Z}1n`+~Yц;Rv:JZ :8sεwxm!T);S!<0 |?1m5.fAh0~X2'OSm%^Qc :UJ#_щ`p}Q 9~hk"$yۂ@zwwqvP=cś}Y~U~3јkY/Y[LrJYL,&T}JK "?7 Id&"\u¹ee;#5qU8%ae1`hv`=^&q;=4W *MP׼*w{̭sIF@y2H%ҢP<QI,Yh{tn0 ~cyLg`?mbPN=*]DsIkw-.[0c`]8;EkCR ˑ ,(ՠ,m9z{S5ѳRd\n2Ic%<+j * M0C{ TFB*VVi4ضr'T_dC@LG6Ɇ*₳SUt%T(\.B QwߑT Uw-N&GP__(DZw脀=o.ᎄD!CWxbq ښAn1$>T[%jå#.k=(:TsȔn (O溰6 c}+g> ق(s~Q뿎߸%X>6? N}I(_ί6j 9:MXK8Yos޾q|TDXbAƯ>6ѿ3KTUOUԜ+T<󨳀ɐHͣE٪Jy[{j$RU[o98]DNrv"e;rWʄ{iC6u'o$Y\/L$>FC~crX .Aw־i$>g 0Ȳ,-U}o%Փ@K֦ ˙&96k9uE#VucŎיc\Xh]E4RP%- GG{cƛm Bwz3K]n3iizRv§&Vufgk;#RM-=t &<3ExȎRcj:>,);}afj2M6ķir8{-QkWu4Y.ۂX.ƟL5@²ɣr)5! [> {a2\Njt5@IY{_11qLw;mE WMQDIP#E3m0+RsV, ~ċezwfN4[_Rrð1Jcыـ,+Zx$ ^?~XRXǝ\02(Lm1SNR]1?Z LI ZْP`\e߁%A&IN$4&zFag热=p;Xrg5 1\Vn;c+B nګ*|9>JݷR] l?TH,1(R3@UЦLuM30'!(G-8ӟUyj7Wd>>rЖ48,+>MÆCtd"BMR~=d‚Ziڎ۪;=<S4*d(VT~vA᧽i9n}jwdIs_ܔ-2R0WZb6o6x{N34 }0=tx=UI *!YRI3)<%cVbl H h&L vkaMJ;ѐo@#Er{يxUfEK8M&F0s-qn}[ʭI򧲛SM#Q @btW7v L&ђ*@/,Sn(6W &2$ MSD16̧3@)k恍Cm xA`oˇ~("5gMʸ(MٙˢfPmD^5iSl@E ~gƸ6 -:۳yI ACK٘8d3^d3,55 /X9nN[k,b70 O&xc>r<>}e%(m1O ' q),OUg `'xK6'Ht#- af"8cA ^ 8' #HQJ|m%14DW3E>'Pώ9 <]l kVilYx,7!#g9g:*JrB 贌 V[+A`[uP2<,RRjxIe0܊'<\-<4v CϬvDDX\7A PWELz26i b>9I~d}\oInQ5J37AVlٲo`ulQj|{Ǚ UAgM$syr헮}k_ *vPV׶`5i ұ]ʊC]plNEgF O?B `3&j{MR۴r`);X# '=bvuZua& ׽k7/y ;gb%(VC^YT/sh쳛/HKϒ_H|FʷC;|)k~# zL`aukMHh[x:japL1ˡ_)򳇠D d8oJ",H 3cV\>:ґCG)*j85?;sdBNc;(ɛe luZ"XS8G>J 7 f!)jJe!hXxz_:~{6>}'/`06 ouZu-mw(a8W6K̈́΍|wc-n?h(Cm{GA3 1>eBF`KU0K5rvF7U;qWU&?GtH81l/{тwԢ uI2YƇsEV1*t4J>#]Dt41AhzFǟ] V;? B4-)l~%"١mֿ/Q팷4*~ъ0}%2ǁ[]^ʫsK`,cqQ\C:JXc^x.ŞMЭ\> =OVhM T+}+4BbGd>nq8"ԅL [|`haxMZ7%!,$hq_ar4} B8~2k96 F[Fr2`MɈVmn:GWwgkS5'Փƞ,ג/lKj蚉d인g:yuL`0U}?]B T2n20Mv!53IzUݔop2['4#_NbU6|GXvkO* /!hȗvYǐiw rɘi w6?ߺ}u%zl$a mt!#C=^w(d5l{;QnfkӃT$ 9H4˹T2VƿWyDc+-Ui^0/) SJuԇ)V >r%߯}sѹN@~p+~r _doygY{Ջ3ZV 8@tF,UC o ~Uh͂v?"a/* R>Rf:=(*׾tϺ U$ 9 &kK|O:AFB?pã56ݓ 3:WIDB<*J< <T\dUbMcuk7nY[?쥖=Ǩ_-"Ų+nl5n>%*Y8U4p.23J&BIб]4LPs rMݺMwD W`\ar扜M #`=39Qu3zm 2CdW6]IRq"[2 {<'ŸP^lޯP ?aS|Wbnl%w>YB=ӨNk:cQ0Y Iq^d=C v%;g'Ul9ۑe:Kd-8Yҟېh8]CоXW@^7ES14ZVra?\6ʷTXp۱½˪p!P@au |N'/i >)}h=E:_L}J<l \'\a~7W`u衫ߛP((fC  qQꂺge?ֈު3Mzg<dE(gkN.&zK ܾ3WkBh(`_\jaQcxr>鱴0$Y]0sl8ر2 _)"@mbxap(W$ѳE%Z 71x)%5Gf'FW "*<:-Wtm!z(9W˫7N48GCN3^>02JYjJ2C +js"$h.&:; -ó-PJ;AkZMOY@\v-Lc H8xFigEG2ȝR.)rA+eIz+p)>┎UZatfSx~I2NSV8h̐Ͷ;5&)$?H! ̡@}P݈:'UtQ>yZg"&*O *c E@wdy >#z<]쓇f3n9\@e<ͳa>h!{SZ%Fg]W74?Dڗ]`ٻnvm{ f&p pP=i*k6Ԍ\Ry7=y2-ny_&!Q_ uՒx^8ӮBȈئm5KHh+&-"ۜ|xB.Jrl)Bn`;Μ%谝E|PA`4xcͬ#,*#nx94V">ܙ#.~K Ҳz^Sx"oYdEӾҌW.R0XbXcbQט,f3ceMG8{wC!0a?(; CQ 90 #HNfw) HJ(Y%4w.W/+W;"}K L.:{Rzq(k;w@y!S}Pr#R>cH4qTa{2jʞ:z,g}ԢlED! ʓ;AʜAh|wB.Հ(& ~?kT\cUh o^f[n{6>8 )ڋh3 ^5x-kӀe6(H^^u)6,B|Kuk٧[Z?k(̱ײSB(Z ecwS68Xx]^"eS̙00Wdn$v?z2Tugī<){Tx~yl |}`ϐw'?"F;9*P1 JnUd ŗaJ -Z)$HP];yTBL׼UtZ q~[t@ޟn<BzaHym~:J/m(Z !׻mv(xZ4Z?;$Qmk$Dl5OP H#G~@c(z hTHj51,scV' :̺Q1`U?Jˬ) A?ڶ&oz>P* h`C)v: Kg @|0D͐qRj *L1skD[uJqUC00k*~Gs]ݹ\_{sTt3ycF>pZ' y=Y6$j]FnLZ7򪐓!Sm]fd=}*h{u oBf~N4xSraysu/ "Ht(7ɟ33S 8%M,|=LjO5㨕rD_ !ً(/Tʫ`K{C\2.W} p5_6q}MtOH[{=64#%4x *&}+xorRO0@U]zkЮ/7?NmF,4RuNV ?C6|7]1D菲gPrLpaygv@]nA<-o[@<{VfSF-.B??BɌT;FJfoLˠ'6x䥊mMDĺ{ɕLϭp=̅Bm_y8WW"|0EC5mݕ]88Qӆ t9k^E Mx2  {Qq>`_0I-;4A9`Fx:=yNjVVU]Sy{Ԡ[btCS-.l;:_XuվЗdO!!{ͣB`yD"ǥyԺq&&=ÊN2S BP|] q\K:*®~Bgi(":|7Hf/f2!޹3Mi,.ȗo,qpm(RpbIN2A![UZG qЍIoWdV<-&7ʘZe|-͏ai^%s( 47g(eZ@YKy6E8TgmA*h˅t*TZ׹PHe,DLz >fw1ZdM_oȵ_&qgS\Cz:plzEȁ383lD%3ƨDἭ ODZjH6O) ;˟86`=&f㨙ˡm_ZgijJalHb{+#m2$4%*F%} AU*_ Aclͭ _iHWGFޒԳrz?^{z>v%u>-6ud`׽ .d^;mL N LJN· x[9֕wݏA; E_KYm[vsUu9oĉzLN@)t?:FnpA݆fr颱ɹՆabMHVM16܈wԳ}5Κ.~,L"Bw "s|}\i#Fd2Ra2b~wF\ ҵ@ZӒ*[phHZA64dFoq_ZEnMSM>5y=lAg=ogqeQEخon )Ӈǧ?w I\"Nuh< :@;Y :5ؠ\;MpB!d6iZ#D ab,opGҙ6~ s8< xFbwKNWj19S`EPoiHF"? EKvZ$ȯ[vo$a VjW `E!u;zjyFGz/Ti2?S}ϠYݙ,E<;Y%Cy)əKʦO n Pj)ƊϞni%s.~skXU~MO]ЂrY 62(ejWR. :v'0)c/BtG1'wD%y 3>Qg2hOVTxS] hF Ҧ{ٷG,h+>jz\[cGiU CqB0?.ܢ(JDSD˜p."2)\7(2OnacȀ { ᰓՄ؇˿9Kb #/?G N/ul1*D":" E 0Oe v{8M=Gű]b$6OZGuܪE:Ī$ 63f`Bі_ jAsn9֤(|4e&.OZ#&Fv * 6oe__֭혬2yu s&T /8bfNj_`H "0naq5bWV3c疔W bWJ!'OYSlJ,!Dm<ٙ+ 縺ccZs|36`zJM !`0Ʀ%{<[:CcȻ'T}៞4(4W97Ih V-nE7 *J޺j) YNrt:Җ\#쑢0 ւ] 7A̔{$ g^(=U2JDa.$q O#å4#qhPW%"́JiH?kwNqX׋v Q Mo XRS) kWdMqğw>hK^Eږȱ^1F(c_AJ<ٖmBU"C͐YzZ2- $~cP|`<ܾ(aTE»C_ziw)h; iub|ruM=0H miZo}5*NuQ%wο<qe t6-:|f VQBV, u HEL(H&YV,CܨoNl:Ú2ǖvV$Zb!lTM܋v%Rbs~FDqBR}SO f κ[C L ;Sn[yZ+)i!E&VǕ* #]v $,n2ՏRg>,I,r0Fk?ePMϨ/{ݰPcvҚ^e{d09Ze+blL,9  qϷ%>\L􈋿J:~Q.}z~ USqQqn?55#la;Pv{|ߒ4@AF.[2h߮~`[/4^zUބLfZ&Ѡ?XhmhEY!sh¦ Oz,Mnһ pr?}šT08~K&-6i)2K!5tH۴֣TѨYz{DzRC oƮ:EmoSRow_ڭu]f9#>8Z쮼FL/_Ш:E]㦥QwUZ݀?[)R c)+* ql] W+`QdI0[M&u,J?-~ B'iso-lϯ]$XMȬQLQXL>O[8;2 ~į1^Ct|Owr Xc0BV9(y&i T.rD_w,?N#`(Va '#nOw K5uLfbvKDcaʗe֕k0%U4!8xoLELfvz}\ уXJ-CpgIL+ H$eQE9z/W\?nƭ~`p~3J4q!G\\KS zZK& Ż+,:kb,UNuWo{}ȯ6{"/p!kg(n2W(*E@uTqJI j"љ׌R~NqcOeHD!Ēh6-:`==W~&Swk05vo,j !>{74HC8y*@9N~O'zn0{7ulf4XB,JLOǏ܂ۘMg@UW3sPk=@O^9@98Pjrv#mf 5i6hDd0 ##N ϥG>3L`FfR$ s^XArI(]WFMe2!i%jx s$/b".О R?+:iPz?r~e>|NgTe7WFPכLU7RsM5bb¢lO:WXǖP 8ci0\Z&o7 ox:\Mav$t|.%LX*4m(P)ȧv]r;s>Mp=oiT^PL53lPNiQT|?VP ,'B 4?(JS$ Pka wr#!hw?jRo{mO/sfZE"B k!O!CFX. -OzU*PCL @x7ëk1h m.yOE@T#ֆ Cci]sUNЊ[Hݛ,uYyR iE˻l)֡9諒RcPToBRȨg/ 35`oϚL2sVlQC2o[:<vH,>[2\-{z?Xχ5S/˦TWUj)n9mEHf;(|[+"Ȏ"pWϯK^{U bwʡo2W AMʌf}VӇc2hGHܭi]# k-KO_"v jrƶ;y5 HCG{ R6vUNm ao,GT5pM|0{bKl\VBRh/(WY{Y}>N9N0AŮ_eg+o8fHx// GYnL0bfaJ:EMvI)z" ȷbȿ79Nx@ 8EA=aSJ0-\V$=kk}VKTgjn}I%7ݧ^( :frJ'"'h$u>Asp(jf-j)!DKE>u`F )RR$"݉aRcMė @4).YkQDo뜜ܕ_?ۓ'@8N C?490Xf?\!C+1n"NsOIs%6k3`#2i# 8'~[s U6_` Ǟ@&5W:OrȈ,?7BBcӜEu@T12;n&KzNcׅ3L8K.<.}q 0W" uګ>Y ZDZu0qhvՍ?ǚxy26f^ XK? )TDr'^t ,ԭ#Q6BT]PjyY+nXUa'IZO>̂)Z|KrR=K0.aG|%i:.B !栛ƫbToU•VM*[V|;<5\1mqA#\"+ׁ6otV[Xa?֗ݏB.-Z e|}ݾ()ک~ ߿xwGV%gDqDfP[#|wV[212XK16;a'6di{$bij&4Y@ ^o} tq!<~&d&=Nʔ@Q=-\Qd(q#)1eXsm:8>[߫gom@b_k6Gq[O~[dz853{V͛mBn,kl{C(FAq+9*^[%jË)vO6= E K!vDO&mMwF=)Gb!U j0x:b +Eex9$*#('lL*$(??}owcFA3+wгbF*(A)H2MY3}ډ,8]r;w~h+tz؉.Rt,nnJGG%m^A[jb z@,9e5b+;s]?SPtAZlek~=0l]I\4)NZgV r>;i(!юrIdk4̑gŨߺy6;P {o`Ⱦ^6HamA\Y+Hz:)\9G/? {o=p.%vTV1.ui! p Z;]Uv}7 p`CIFc^5nNgDaуĀh,+VpI;ayG6#X " |?.vgo!4}zߕjc3V˄" YkZ'8gކP5fZ B8Ŷ;y~b) n!!ܭRb_ij=s]>KI3UN(6K AqZqD_p ɣf.lmP*kZW]:Kش;tV7BIH'z=It<ҽI"m.k9XMruTgcSuJ+D#?X m9! gf68@>i+i*&:_5`Lp{a eeޙDIK"~!H̬ S,ptc< (bZWqvV 3 ȁZ$&|ʇGY uSEޡҬ#ZlߡDp8u!8+tiuo+^fﯗ?6q[JΡT% eNΝj=~a<)UGP!HT5mjv_/p? qԂlrXglɠ*0>Y٩7puS.3aM v J)o~0W?vSk.%r 흢U.N;\䈮+~R*Dl5-RQ3 gwg_vF}I訕i9jQ_bT9隲νu#@f/9/yx1aWYUfD@.Rct&TC39 -:*( B6)܈8܂Z-%/:ka;&2x^6+~c泞s..5PL"կ;EϓDŻpWMt,>j09|+בJTpЬ{2*AZ3h冘fIiQz!堒2JFmL>hC_|Sv83,FYTӋ Sʇ;6,(;PPr[TZ6_*3"[`\ڶ0}nQː s!f5#m[| ꃲd_VPPd`bwrFєF|XT~ꔹ^R+ǽLLѪV@ٵ5h18g6aQՠW,f)۽눩Cr)|pgM{ϽGj3z7:c)ݥi_a7kҽW俀%p9^U-vD;*')CO^!ƛZl:\Jr=s&]K?0gtbQ8rNuM'-Y^FCK0 MX&r_ϔ=-=4jbD-Hdo\кze,ND!(좱*JNjB(ߘMTol<5 @[d?jn(Eo/wUQ.瞛sy _Oh op`>{N>j?7|F'rv;JcaSIJi@E}hIwPeW[*9V^RL9a/#I #{Ao`5,͓h3 Q3?`7.Xn5%0htye@Fe٭n'G="1oV}Q=KTp@F"{ G8sg:l LiTW Hj31ae5!#?ٱ^bh5e)>nJZuAx܂/}x kkl)_ߍ9xsOڵ j5X-RP5MZSU9>kZxQeA2 -M#Qyoj-خO4 -[dX* h# %Q#*7f6㟐1h:I7V@w<:{[̶=]D#ے߆;lyL;l&vsJD`) dy@Yh4u'`ua=!DRQC>p<2!lo*MPWcI:ϵ8*$ᐙ4a/ҙFP%jEH?1vd<62 8.!MӢOc\s6Ů FF{S0(_%Z!xd=W垵Jcdv,mSuP rF#4.f[cCcE] v$yMX?D^aBkD8~ӓX#5k/2Zuzhf=F us@qӺc~eN~uG'dt>yK>jATb>왤1bam>Pq3T9$Wujeg;S]@DU}y=rqw 1s"PTp^}xYT~)uCȭY;cn47V hh5."˧_]|')#K?q2Fʳ+0BQ|=e ;K.;)xNF2SX+Gh1ntңBr,'ҫAlK} 8{|`,T K?X( ԕ^d:5Ԇ $#&: կW|,0+4[(e!`R:{O6$r [OT}=K|mPs{ 4ƞdG='vø @v#%Pjpˤ ? i2>N s3w%ks5J鏡]6.M,pF`#l@uU);&C4u* bFCKbHFvIDA.|r~F-ao`V9v'b%JL>r%ϹLKSʌWv3Ǡ !xF~Q,܊pҴ+;;񢓆!:a]OFL#ip?Sѥx Y-@~+< lՆu,$Dgv}RY,\3!>ɻt2$jn<(^Ae ]V)Dg^1Eɭ vJ qr -ƻ): LQ("L]I ?Ec*Z:< K=i:;vDR~Ƌ¤d?l&G\6x6\HyHUDik#s_8Lx |),pU?yXi ?jw8قEcŒԿ&ʶ@w'?&P>2i˽IK cgacmy*5#:ǎ-ۧh`>,m d{dKUc3GyFEy=|TکF/ 3*`N'=ي.Kag6U9@#pL"ޥ_Q Y7=nXf4bKs WL _qR7eTELhҌ*pl UennGLXGr.^bCF\~4G"BTe-& y䯋d>fk'e^±LJ҈s?X )UkiFy57( ( ]"HmJʜPѵʬhw4mѦK UHەl)/%20ktō@ 6p;NgvpBx\gЄd~l.[oXǶfڍp2P>[ǶRF1|۷`b 3Z.<8x-yG2k[GZfE^pxEHH:B;jX$iRfǺ礐 m/E~ȼ/2+֑lAl Ps)avxxY B}DZAM8c:'U3~|'t9(DlP5[hbqz#Hz,k~TKɬ{s`<ÆBAxz瑦pHE> p,KS hXp+qzPBOKB*9E9"9h@8Yde[j Q9'qKP2~+[@x$duߞaAoE<]1q/pȬ0 tGisMG G݊s|? c{̳tv뱱Ό+Q [u95El1]Vp:*E/UWL%ܚCMB;.4}LB0D %YE2IK_΅h\ˈ;J;Ǥ;D"% !)=SU%+,O!T!g] ":7]/x߸(~qݨ}C%{-?7kI,Ѹ&`@ ?*]moֱaU >HHGUs5ח(_yria=CkOt8S=W)YoWw|H#͢<"eQ9)ڝcRxk4Nяbʩlt- ՝fHWFq 5S\d^C04ChR uvbN[Kbc8pK ,hj:SCKTqQVYՋ}y~[xeix KǼ΂&{(T*:;&}mr%t -[jʠ//qgqJEo%c벴;aQKqԳQ*gzbH?ߘL~UjF9Ĵ6<aƄDheV CQmu& ײr ?J с_.-o[1Ա&6 da # ͖t/K{Ϊ~i 64c$B:^!ݕU%7LTd0@P5न(S&422忀hpzɚa]߂/Fu5dC]b݃;ǛuK{*x1wF;y!i6`9<F{26 dXx̝shy6gB z5K)%RC NL T́]%+&1:&_gؔ4Kewx( X)>\0hk&N|q(6O蔶P;l/0 ?5d_#$oA+E?,4zˬ /rilB]sGDN~1_ᝇX͔zYqNq 9nB> ,`:/Wwer  Rh2xQO7",O$1`xm" uQ[3f@QHo4Nӽ=TC6TWysH?8Ǹ̕bZ?1슡 ꧼޓuD)&@\qGJ@iHS/JyJmX1^<ҳځwċ^v]mtҜJq p==ΓOcݻ\>ōJc 57LT46hh90;{hrX?ڢuƂ)<݇"U<3%@<MēlҎ!(r+#r#M~2AgR ++a:$7z_hQYsFgS#0W86_[ v7B_PFO03G5YӒ|P%CV0V i8\&ӓn՜DZ s=RWf-̵E*f}uSʸ C{&%>Vhݹ[39Ga1ڶ?>9"N~,yBJj?H‚9xk! d)vIdmd ( 3lչ!RdP!#GF1c= "1Q~.v^6I`!ݾQOy8SjM‹|^-& } B\z @n3 "7G:5XiGBhcHf¶ӰIz8)XKVVmN S*mʲDOqvq2/y}Gjwf' Mdz_w_U&]}4їF]elDzy6߶P3TϴFc@|lKx=H[G`"̮)DQYd,qno:pq K]^J@=j(2:_\CI0twi>hW[o3P,=Ka~iJU+8̛!}]%d.мCR8,f<ae:&Hs쭖>in} g7[+IȏG#K ȅ_EUMp2KjMaأ%b#r4T.!SRS t"WTc3F@_)ɛ(Ì '0l%֥zS.,wz 9—FZE.8j,xawh%qk sEj)(`sKF+nn9shN|uP/VjP?DPFM^/-PM8۬ &`Q`HgQjim :'=ػWvȒ"yrlƖ(0j.NA]~ƞrR7kU1t#wzY8Wt֜&{& ̶q_G*[/3\!e@u:-~NS&sj9+Z5/֏5c 9>L.mL%s>Ʉf~h!؁fdtSȌhV,5~Gl*ӻ/GdΒJ(刣Bn٬i;9BF{gAQŤVb` E*z`KGfvkp+RBQ[ "bxAܫyVONm zX+i0R8(證G:q&Il:pkևN'7n%H`BRENw3|} hرCxjYsti =^;%j[d#٢8 ݌%8fKV~gGo*NFcJ AJ9;Ʉ>I+sr0v~b *[I_p+b*'ķCƅA"p/viW%"i.`6X+X`l6l*F_?Hv=Cq='Q\ؽ2U6:JM۾861J=4KgbQ"Ud|ΰI2:Nrs[Z2`pA@* ~R_ .]Њ7!Ǫ("F!mAtGmuB/$K&֒LOI4w]ûPE^øjSvZݑj VڱpO=srkBd_BEP8X_NӜYfOtwV\ȷ#^yN3j٘F,4π p1xGүBxJBfd#tꛏP<zO5eXe:wy?zb9Jmtq8Cጸкn0%!EJe^][/M@Hw=bt\CJ4-h7yr:bv@K΁/7GZ9tM+j%U-t*گV1"LJ鷚=BG#Del5H>=Lo=kJͱ'H+$ooO/48ڣ{g$>e[ٸC%SĦb9ݒեܦN0 U'VL ZrI?o)ZdXv 9jITt)@U^~C&( !>6ϊp;?3sN.W+Na:kwxRg3?? ei+(&@O0Y&4#{x'< ai1W#Y|1aJ\ν!6 e1µfoOywV@NEf1+ M( /*lOVA o>)s\_U6Qdo7 m;& 6C>X'3Md!vVvA43.\\@&|d ^nɓ[[vc)`8Ś5lDO2B ɂ>ǝlcf mG.pe,pex`8*%[eʻrY<&:xrBVx`&:53.|Tr+i6F:_S$M [~sL{f_#6(`>Iek ^Cbo-p 1=PR)zAhN{Y+|03ga\XB1pa'^W=0,-@c9$FɆ6|$Ⅹ'q+q=dt[FcB8یB=ZXc :p⤴RC~,? 7QGKK=#crJ9( 8jGW=ձZ.RZ/{HtF:ѠRx@`Zp֌ |A29? |NW#xIHhH@ oߖee`$s%v!u4-<Ҕ/xr^EroGɂ[=I~U X8~Q=vݽMA5C~}YLV5LoR+D*P-@D%D5rWخHF}j .pINৠU1iyȵHHW bOR_DzR]@<:&q" pr. c+C1kB5}TfY Kȵ kփK8A VZmBfD"7:Fj$ i% ÇxbTvY.]A!I _hM g"`!U*@Ss];sh\X [?lIꄱE)"r#+{<ӫ2I!m q'1 E{}'^3̏̃m3܆?vqTf,3IncZ3%/=&U)狉y=\ߐ3]kYyVK(nt5TE'm"aV;bzlH$B FgPA C3iW# ylbgTYMP:(>:|. /'s>٫Z@ {Wu%sK ; yuE]ƓlvsѻB'i41C9_Id;{sMЂUJ,Q-]!o-_W ,2NH S}~-˪#ew0jbf,$}Ū`\Jr/X~(sGV7,W6sҁfn}֒I~#V+0ZWGOATnpwQ)s>{C'PyP<Ȼ'VCG˓BonZύ.jOͼ^ aRV~HfIF(#? G"Yk<2(|ll$Mƣ'twx$ N<"'s@:%otn͑ J3=#YN~Y'O,‚E{U~%Os2-g_؅ ?|3 "/D >tr@, pwP~pMӉ"ko$HzS(_ =O"NTo \o`[JJIkӓ-09%ȥԾ ܺ0[KsSuJCԥaFVތv/D uyiU6(.P-ڷu*.p=03(NjNd\L,6%3mWHҫ%xQma _D؎sFd5x5Q+w.45eIpAj3}s=jY92 a4-mT̬)QQ3bkVB/ض&T5[-?1S-<+jao? 5BF__"3(z4SZVZA5YvO>(` 5&wn3cv w/xj"c&p O./׉੟\!&(V95;MkwTj!jV;HCZ \-}x5x"` O֣}Cqd<P;V]ck`M">K' :+m| }H2kͫT#F(|_8hUZU#{ZzO>O[w"<[8HWaxP jн jQf8qr.H֦Phj5>uP_D>ҧձzThKHpӈFdWO !V)o ixW]>wL% PjCY\]77&Ǧ/[Nb>a?Wd dfĦ}ckF|E< BD`i*MGRk\DcU':zYd e#ܛp0x[- eX0Z F%P.rԺ*YR2't\j5UlO DDke#.]F lPV}׉Ddm+}_=+v5b :rKHȑ@C2D!!AjU_t*oTͱ5A{upĕ1B*YǒX}U(Fybb{Ip4]G!`~-?]3ŽHv a<%G@:ܯw1m#.N\yFD0|u; sRʹRڎ\,4<$Z<S]P´u E&^vw}`F4VZu@ԚИ~I+&&3f=s-6\n_nOU_|vr4A;e I5%]0#q3B(1|4KXT %ѫC4XE>>>38kFנ,t$KQv桝MP}mZma3 ceSzuĶ#xine!=Z/k*U[紮އJRzئ>iIϫW-39FPiBK(n ^߉mB`1@gB.Tjp wH,T33ժ-!]b@Ykc! l&6 ")بבOޯk PUMhYQ(C-8٦m4F3yesKo"=Rave(ϗ}M8r;*i">Jmzb.9G([=r`5Nq3NBUhy zfSYl.)v"Ff= R^ψ[wx>)Sճ5t"iBr'!)fM~2v)[;_eJi&qq$aFpIjls pu=}bL۶D-n\I+ĥ'xx{rw}!vO/ە\4,9ѨVmqd^rd{IY!'Sκ:fȖ#ׄÊ7?'  y8Go:*U KX21TQũ1RO=EeqddlL.4$@ >ZOՇ$n8dHC-)S>0'CY&cO^͹݂0_,@?DF@.8';YO^\wOLhDaI|*$%Q!9_X'V S0n@4_ ec( 'Iw))WF,+ˬg 'Mq&6S zF)vDFL;(8r X;7NI驪uPz|a_93_7;ٗm*~W;d=Va'TO[(ج(p90eGO`~geO@pBC'$j}]n^-0d&KW2?pJoЂ@ʜlL 4v< z~͌gw`5?_r&ߙz\?bЉ]Y#C`~hr Z^r|hw .,M4J- 0Y =m[=-?=v.fR`.MVE`]-bⱷ^z<;C`R],puI˅Ic闦2`z..r9 DKXxx|% I OЕ]JA#nu˜>]Dnn)GhX7ӊۛI\$ݭQ-nnBM8WE`IMC͖~y!ح%DA-Fm;AN[xNY!Tic/Q/Z@2o}=? M %2vp+, Ʈ̥4}_v_/I0K9'5ʑӝ؍*m${ ?a*X {XZq 8B^W^,P;7p= ` m>"_:ݠ$".&Za]-JpыE̦(xFF-ι!!&&w҆?Nka*3`*6L') -5j`o ~cJXHH4乢z[P !IhbTy0ѝ! p{Ϩ={iMirpaWMm _;K쟛ۜa: 7~ cmR,z6? sJLw}A*fQxNC43+BhEF}qu`{kXw|xJw'_lBI:D?( GZStpe^S>h3>+NEF W7y+~>S L#w&0\[L`qfjqeT `zzzDv\#ၴO;ӈV{-=jfޮ: p%C7Rka%nQ8fK?Hudނl_,BeA[@(g,^c}_'!)vbz(hȂiAGˁW"[tCOvS۪ؠm2ުn'הI (pu'0p1qAD.˟ʏ}.x2g}2:ȔL~ .Nď_Z"]1;d! HDCToXl%U{QK+xN* }@V;Miq_uT$uZq@[&J ghokUy~ h J82=:WF&i=pʇ(+sCՓ1y>LQ>]t\cP1̪5dvhwt>w"OAݫ  +JޱҔSRfPnߝ3^3Ό~zh-41dM]Q>-P~o^Z8Y{")ʼ f9@bi/$L"`w*N `Iv%n<8i5!e<*3:4@Gơh8nDs/^ f?~AH+Kxzh]zx2h=iW},Q XÆs]^bRi1*j b̩ACkc4fsl yv+V AJe{ὖz5zLQ;æ 4h܆d, I| fF؁Y;n%sA4jMRTX)?s}tE'IHQŌ$цFs6h(LpZ Ͽ$;PBIRqԓ2-5AzPNK22vATwjx<Mv&!")6P5fUiEbBRScAn9Ѱ>̝`Eya-F P} ~&t 3iLtJ_i2+ %+3=ʪu/%YTg՞R㪢ꐇN5W) b!vJK1w m)31}Q@oP7-aȉ+ k82*<lɫ G@ /oy17M&Ty.y柂d2HMb'zJm7AvW"_ iдxSۭziyl^ehbVYk?9=|!Ja.@SӇpP8dcna^GbT0i0 {"0=v^=*zELNbU7BPlCO}ad; KQX'VE:zRGdVR[mc&,3 f0EYvTÉf^;~* $16@9xs5{ZIcAӷ.3pQH.QϢ o9BVjNn1@;:H[D*Y AeGa ZMK[KR-2`l R^&q͵{ z J 5p2S~bΧzy KȾCd^2,# K7jjgg6Zhs^A$KcD6UDl^ѦqlW=^or6;%tZlm|D9 _V;ďlo-C&B>SCsB ٢ʴHJ:m/@L*Dxs =7%io矡ۣ8~9: X0m {~lI2~S8U;B$רS#1Rޥ>MrqpMЄ4 ]I!L}4;[!>8P(7'$Q`ç=_?u_:?Sb FʎFP49O]'°qI #ʉSase &)o;; _0C5;G\Di~Ajjje gbe-RV }i Kyc 8fH| Ηpuj2#ytxv'dǻЊQ-).z)L $~T8@r$'w<,2>Uib`طwέb;f p`IAzΚs=/f/*.ϯ^Z="Hx 0FfI/J sTn%cŦf#P>ں6 t@o:۲"6q7A9r?2x'DI _Q'8շ/eieA xUI;2 j:7~UG~$V~6AN+Ae4Lɚ8Y]*l{ #~y7Ih5קp2Tuv~+r:~MKs(1a1p=L̮4-Qq'QOVt69I_&G=#wFl6(: ߆7<p!"aA"j),xUN:v4&Uhz~(<?ݟRkc c hX=[n\;s$6((0R?q h<]9L % x7)_ӻ_Anထ e>RTg? {a+1Ӵw#&fswxA3zk l10pA?Rt7{qwwܸ5c+h{iP?A9JO7Yp,Ush ulhmTQn >k8{BhhG;VQ'b%Ԗ8~ȋ3"T))w͔c ]uhMAgPcQP[N/i9f8O&"G+7 6$O.$Gb2c>Ncż,?A'PFǫC=pKYw&\)(Mb@z qX?wt_zu-D2%$p򹬣y&[lІ[".+/ 2gkK9"U2u,&9vuI؏YcpIнKVE凃dvSca ̗]!\դ#m6R"CPօ7G[D]=?+_ 8:UriPb\e]c &4lrE2me% (ZJ9nǧlr4紫X cU.xig! Oex``嘎p @1N&;Tȫ#4ppMWIW]ws*O08/Ԋ'=&Ez``;E7܌B:eBS,YVృPAGAeS6|&B]%Z^_4O.;p$b(H19rxWk -D? Tבz+-jOOSqz/ق0t 4zHPN:THl2sW>PX{6o=<*z8PE }"Bǔ/3B";Ѹ|VAzL?D4m2f#xΰ:zKOs=3YڢvA:*toic]J5dzڵQgle瓈76u9=mdAhTtVsqWR 1U_b%Өt~1U-J9p+A;1MÛ#ݷ54-Wv'%?=,/ 3=2'"<~+brLk_|ߎgk6~׭zu/ Q1rqoSySLb1XrwR0?)3L@"|J~N[_ZHF;R[PaDeM3 L,mxH#>5){ k {H$M 8Q5 LtD"{Gġjsߎ#V)yb9Kd X_8 013 S|5?iNOQ!7CK\Ԋl+6{:+fx\Wa6B8o?L"o癷dOayu"(t;wf:ql1#=7zy5֙WpCUF |Z0'˿Y&uQ9ˢoH5 Gҝ<1Vƾ3~ɣ}DfDlQ󢽱 j0p@\78܊R5(2>KǼpdHYQpY8JKӴXA(I$Z}gX8Ƌں谋q\熇A-p'_`jS?YD{I'*EKސA lh={co¸Kleaͱ_6nakE59N=qHr_.tҵtr:KrRMJ3W[)'{[)gӺo Xdqӿvqн d_?`{km`9b{kKuo'Ya,[XIBI8{B]xߑfu ^ `$ 7H .}o0k7*ݑW5 HR b.,[ϑ Ÿ%b~Ami~Ɇ7/AQ7aҖԲؽ>COYdtIO?Ьo#2ٕ*+epXob罞|`D1>ҽn >I\RѲ@a830&Q ;ɯk(h3D*T5 pD,ukw]aTrA3mĒQzes]ϪqLtJfs2Gf9'D'P@gGU4Md |>JVPXVU|dx +zgbY*mVbB ;+UmhUms]\Ww>YgX $z+)ʲ/ʫ(ܯY o_d{Qz3fk&ğq2ߪ̒^>'/HEs~|2!8HA_=mk!"Cs"`Lqk,s}k<[ĊqօnT<(rǀ~;!gwHE5t0yTݮcmN~e\ =wȉ](~{1FtkE|M 6i-AZ>w:/Ы=nZMbF(6WC SM׆,2mWì86iΘ2l2 WݼbmZ4Kܫ# -6eN<͸M jjeӣr|QM&:%1zE1-uK5]o7h"QK`|[ 2()Eoj]njE W8;)qMHsG :Wt- 3z"E#1 ʊ{t=, IG&1 :jQ|3Ҟw^F %SU'7)έڵ)K%7 Pzj0w_9<~}#NL 6z"Dm ҷ%YԱΔa$~i5?- tEIh' MZz=DѺn:cChgG:r `ŗ1x{PaOWCT}0; f,d[8CV5T{DZP nBֱ6<450c> sp%@}N q+Lpf\Kg]"/ηeIJtǮq 9jgc<о5KN醈6٥'2n}D?=.m. >ŵ_#䇏(bsF@nIl_`Ys0ҧc@[A&|'Z/ Ut64~X-j.:՟ R%s#Cd˧vM>\}f-4]"](&"LBG5Yڦ*PzN[`]fVYX‡t(B-1LRH_K.SCq*z=ϱV@BbL{y7!zƵd!<OækbXb:ojM+[3(X3^$3_V6lc?o X!y%'9jr^t؟-/{G|W_&{jxz/` |,qɢY5SJ7T>m0 ™exҡȻZUbH=~O5;c7]I8kwhTXDܜ' u+nv'`[ |JFTX$#w(Y(m$2@ޥk}և;-wbiuHG_)VOJN/m~7-,)(Λ\f3CaDqFCAE Zט18`YjwOUkו? szrщ,pKA&Po1|+huX3k2pRmja& `l'ЍV_JFo%Ehi#@}<^wVR2 BYW&1;[H-ѬpScUidețf<֛Ö`է2>h-}E)i8;P&(i3K 4|0s}OwZBIcuO7&CUzi0Mp/mqPuiBWCloFqWz2B V>9!Ɩ֕5Du8r!} r埔15sݑ緱Mp^\/{tUd3LvJ}fB&0\&XɾtX`. QH g}絘,4LWu嚀XO>+7`T}Wd'6Jt$8 maPSܱ2sX€y)TTݯw+%9k KӦFKaC*1)9T0Fԓހ5}My{X ^O/am4 2W҇zY߈Ni')ĢvH*ShP@u/!R JRP H>{^x['d}1-fWB_|VӠ0%Ik| PjWM%t'!D-[<]/.}7S5;y!0\`{j!*ļ&԰ _#ckJV. 6ni#\ ? {an=>" b1YPdiꅨ~&֌Wհ%x= *'د n^hpwknz"KL& ]/wu:J0t.*2  ` uW\syO55e=ʞ!WG+zÈp'Mjqvp!#[N$ز{H :_Yt5E@m))h1<Ј1!}@08A|ŰjkE>]~,f D|IŴXx}P&I\N޵TծFF&U!tx\gg)*szcVd\pX@k6=-7zem{  -tƃs>zGp1L{[SO8m[VTG 615ّfDJQ;c5=3ӳouy11w1ٴ]X{t(Ω(OoڼOOoG=%W&U)bv։;aK]^r/ +6Tvn{'HY&#ѓIE*D ad ڊKܦJ[#+xiq=s`S 85p"^Ul&THY#)h1&EP3UJIuY;Do󡼡X\xВG-oP,yz/;uUqDSIoq5ֹ,>B7^s Sb쳃 Ѹr,/}7hc̀?Q $~渒=]@E>K5MFjI!;_]/qg\ĬQ vE0Q" n1[ʛI⯉Ȟ_L|,ŖKPt8ڬMA"?p/C7hPN_;gюrGMBjrPm\Q]\xA%с쟻e>Z_7ARqCd3΄o/I t ޒhʈtp{CK"bI =7h08#F.3T*/ō>Yx*+7@"--aҵcҒ& ׬hVܝNE͈Xg\,KCYz‘ey2?Δҏ82Cj2ِ,/zw @C-n#slAdY!{3mG+i-I\]AAAnCI@5e b^}8̧ҢF[crf zE-P1P%jDyDyl, d [+ś:UqiYδk 4fIr[>.Jg:)9zElyX D^I>+82ɛ>euX% j v[smYA.Oyz`Ů%}}ъ퉅X#ilY: x,c}p!m<0Sշ&6`@.,/Q'"S ϩ*U":&.1 316FTS,;:~u"9]6 fwU-92 Z潧Pٞy֬(hiUt4J\RYonv'0b6j \MXx, ȴ*8*u^A40 5j,J~&wR)S#H=';F8Y_Y}--'vϺ1PDgM>v4XͷyǽG^CW<_ I0NCw*\npk${zI>r쓆B 'm;Rf'6kg$ebqp˜Ш81~F˵vtsԋ*J po<7!3@qJblǖ܎,LХWՁ19@b8 Y(:pY5z8?o,k^amHӨj(I'Td_d18œ{BڋXĶ:es800-2V0$ADl}\Aȳ?}>sa^H\RvoN:U$)"ʘRd Rݱ^^=4Wj$B1vUx/UH\Z.# s;)8.00duTlR>hyy>c"b&V eVPq@F+Ҟ娓-K,?>HY/^q9H*",CjrS};j}7ϛU/p8C3fu8Ժp3y:6KyB0KM~oEק$rxbNL5cHKn҇$/G G\]Xg6CPbthCASvr'EvLJVUw|5ƕFG&o[qS01LBF 37gyK\٪)A)>u<Ǖ =/p~6bFIታ=&Up˱vXELŽ_H!& \ǮϜ<GRz&Cݯª %@W 'BxiV7LƓHK`%uefY)@f %Bc%ɻ e*&E=[9~N% Hjznxg K+Z钠;Tg $2ɚ{0i 5MhUqMJ FhB'ґC.UfaAhjb. )[o|z!FQ| M ;֢7^oHW1Kz3Teq&HI C8;w,DqD}ضd0B0#', `%;+e+?4*(ly;@^/H-yEYUM)$YBP5 m1|~N z.Y) ޢ~+ yDm7n1N=)TmLjtnU[|KY#qFfMy3 v\A s-zs3ӈ.,۔{=XD@,(Jּxdr{hKB|{DUY@;\\qxZتwp[1:ɷ} f<5ux9ZE<֟7b_IjZwPoև"^{ĠUkPkRE'J[b`N#%MR½٤.,yfE'Gm Ȭ0%զK4tϐTGusYǥs&7?$W[x-[+@aiRPYJ.xY[''A΄im؋H|Z(zp\8nݟGo\}7\ca;O$VKcAp-#2hhJK{ L92x"ҍʧGzbK7uIk\9 ~5~NLIs,8KN$HSYO0 E_,}9(b(dp=}Zy 8 8 *W¹pXVoCt|zi.̷HfѵʵYPvD~?Hӫ;M5hƨeiѰ:#U o{YWa#? t=EvwfDt N>'B.^eO#kǒkYi80K;Aܼ{mIG@~'ժAe.TDLyY@@XH"i y,4NUnf<:R3*}'H'"HsBuܼ;a d 8R f] i/,0Y+8lVx`#dd);DׁŹj7R8ˮH(,IֻIby S)?MD." :LxGwg۾b}FՄVz'߯L #NT "8a B +v#R ;lgc%T:A~A IVDH3Ag> D!x߬UE KafH{t5e\o8TnA#_߻[ Ư4l頤1SFf=jBOȾ'CsH+Nn pz| D!pe:@.}"JVv T}"/JڳmUZuifwܞ>ǩ:P-#@x'QCd]w/dP4.uQĝkSF$c]J-YпhVS†vklZߛ6Vǣ6hN3v]Q0gQW*9,,0ZRڭ| 8Id.gʈ~;Z {-h̡Ml|u('PxMv|igvg@v ,1l_p&)mjصMSiK#2ypz}tO U7pku|nN dfhO C0o]Ƹ[^uw 0a4[hcy𻵝h`mJP+ f~_ - [~{uN"mLϟE1J勡~kV\Ѥ4qvu*/yǡ1SUڊaW*˭6LcqВLwXvm$&<T#1{bT;[n! _U"]TFFs<[nشKl4->F|(&UZO 8,d.4oynE|+#d7dhgܻtB Z?`2CҔ{_0$m<c^\ JD $B&0d-41j4YZ&R EF|nqP 5,ܼ ?ea@ rs[K~-^ɍ6=4Ǝ}X/ÿ.-GǿACeLc-kS˔LθE3\r}(ߌf_+^w_w oܨUf9e{Q n7ȖKl5BAׅ>ʜT~8AbY&7N[?zv`S`HvOVGPޯJQaC>ǜA\ԯhMta]iVOėjDr7S ȔYJGI=;-BuEu;^7`o}E濻S[sn7Ɯe1=f [eoޑ'ʁOHt-wQAdK*h|}XqMkQaEYC?9,ʕa"}:/]BIY.@LtA\.䱮 V \o3AN3i!En\_N\?j 5z-kٜѦN㽌B T#,]2"6غ4 5mtC.:|Ul_kID!wj!Dud(56)!pL \Wdd2l(:gMx61# a%-gjL[n y8##.qީ7Nl׷{m,v$2iBLV+`tRa^ LXh>+ss "m&=7&M[S;<M}߮GMKᆔQ1/ghb=\+yKNO7Km3(ݴօa cW?Iw6TidM~*XĦ͘an s6C")kp^ɵNÖ.ʬ .Ԫ;͎flx/ZS_ps#,~E7,1&ԇfz"rvj\f5b"2p JTuNi91X'.+#~)Ys38Oz;M h(8A6|Ôєr3{"Lov)uNWjfp&6#Jql(#0Pvu@ qmi>MɈmiGv qIzʻ̫m.y RrB #g\&vYBx r\Hm-6skpv,%?O%Vhmq"!YIfFVXrG. a5!~E"| n)ŁhTH]'p~DL)* %Mպ ȓ@\φJnA H1X#jSTPV~}lu Q8Aa:001q.LU3'Kh) cb3 Fi1n1\mFr^vtLc0[d RT ( o٪axeT܎|x!.ϫY#W@KdQPQ"\ɞ.őԻ4a0rdcKqgiU(a< Qv`Ϣ\c-|3i#6~*ã}Èxn)yHҧ0yo_ʼ8.0U>(R ēmH(YhQ%P!h@+ՒFCmr ޝܟp^':WVm|jf΁0Dǖ!'&m&Դ71&9 |5@y`3J)>k_cmQWǿk/ح;RˌKZg0Ю-ʪnq &"c~nBˏ xܸnvifUIs`ZZFpIsi} ulY˵m x :t 29EUDaVo 4 m*ٲxdåY(Ϗ,əPOdbxE  A"ۘc TP>à^fAizd(Ɖ=nKWc7 /J #j[J(|Tt-_lщݱ)aE-!-6=_0\7(ȕ/DyzK P߶gVz /H&i(L\ 7-?Cd'~+ogW̎}yXЁ5=ITd]7#Y^!$UuQ\L,/_]؁`%_a%*ԬA\WESj 1iKnٻ; vڪn7tL ?/6A3)B^`1m>=~U(X#$95=!CZfI K>u :7rlё%Uq] zl U=u(UhjH7K07bRcZC&](S)cir؏n&EK C:UjބbCWkgK 3UQRd&#EGi9C uxk"Țy .xKj2tVs:6'Ể/GZhhKt_3]QifB&L\f ,)#<&*-܁_ۇR   Lo_dS0^~NES?Jj ``ˠ/T=lLQ{N0iCnPE;[ [(۸7уo'A0J3r2#1?:1, [T$#p߄Xޥ/>ȵ)"!(ݱuk4Fj~d~v V0Qa^pܟ<n 4&KP7 .YD5?.{;(^e: ZO *+g1Pdieotm!L!aW }Δ+zWvZ(զ\r9A!'|WVev&J4 wWXw@'m۹|Qʹ(A;+clNDzO+ RgD| 50]J6Bh7S?۠u.a1Fna `pэ?dSCLݪoMQ=C&S}ɒ;듗әK P۞>#!}.D/[m`ǪGZpD!,bdZP l7KGRH!cr6|SQr5)nVvZ.C<9r;)ӭ(;h;'ep6Tp$ZyQIxI,K`0\ bx[6cp0n Hc#-tlVt4-K>5%Q3 +ITĤwjYIW^|Dbt #xQ*` 9ҝR ?yN'!akV`fXӄ z|S昐k8')&=:I ?Vu:0Z^A>7ZSa[Q2x9.>K@ nvȡ4)&;ӁPaV}%qb"on{5x63Gw6CIIn5Ek_j'0Ai%gVU0)o-٣ME@r3V[0^RLpTnjq^+˰/4Y H@ޒ"R,hj{veswŬ@Z@ﲖ/v:;^S\_vڼiFhöP FM(( IB`B]4ڃ.(:(S.,ݝYHKS`J,y1XGIxP 9F᙭ciR*l ޳Bbe>2:#sĮn8?)i>kx#ޡ&_~E;?lz4D ]W9v T(, ˈE ;=7VK?t.gtȟzA_剩XB' f2AN{,[ḅ[\AQwx?)jJ\)i?92Ƞb$@ޑ"XUj ɊYƺ}"DjC:%tE4CN4`t RX_Cv4p- ̈&>ZfKԬ W7Tћ|c9&Ws}ln[@jU7t a߯k zoRu1Y;L TB۔}DNn (h/lL-QuM,IãXW̤$^X1S{}v]ܨ= 8!d;އz2`]u\YG?r>o2]>X|sFzkw o!_:EԴBcUtX@,=18 |ty4q,Tg苴W^jV aO` +5 /gӧ_YƘ+C^qfP1qۚ?laO>~ŋ᫯%a`-7-F{~5<5+bS>+L=[ݫnK +] DqnDŕu`1wןmѾI!Ǿ,$?M ˤnm[ʤHtxaWGM PD]@myt=_4բl{Q1fLiO=rEDnNpwG̺xkz(=gz5*$#CTW[6dwOXc$)iZ#OFL"x|"zr˻(.|BɕAsBI7>1; 7V<l5Qj7&~PG.~p̛ ;b41X$E| t,:N.BF.E*o9)gSuaA;Toi =dAƋ nLu*["ѯb;ZJIA}-(,7IIs/-âZT5Q3>o@+n BBa]8w g5p Hi_aN-^rLNT,&J{<M7Ozca4'ɻ<TPw*y]'Wf#&iJ m~&fvC3}4R^}mZڂ@' )~>ZO*^6Z%Uch-*^6F0rC){^_ga2Ξq|'Kraӡ( \rXw4$t-dO4\6D)n*A]*z\0}DmS'NTrgOk:zUWLOtWܯR%!sYDÕcÑ ).,_'CE`H0|~V ՘}F8"ɨe\d5}Yģ:tOmDSg!U~2:9(1ܔm iw74mGU`l j[vac;ZH젼. SC%s̰<^ ݂ 5d. ~ho'Duj9U .v5E19z,0?BQ YIk;42cHy]v?>W5R jR c86RuP:jدv=QB,{""m=@2[`y$W0i\4 W',IN\;!CV /0dž;(Gu%ӼʫFRj ?1m_ab!XNNfg2UWBoұj3NT3ħ^#ŃOP.[GdClm6Lu3̈nb0r> ḠW0Hg^TnKGz MXFjun ,L |D mP{uX3_sG%HOs- hZX|V%1 y[$ޱmq&[߁*W~fhv}PXe$nF9qsOGT.} -_jfkٺ_ qOY;gmIB<?/8D )(%oS4DjkF< gݪjhZܲ.4IҴ&`rHLWp2Roq!R$ Cx 1KXp#9S4NsbX"_WM\چ>1j_ i-ionM_><Вac:s{uMEgkX YȦ ;NIc,|E!F0 yh3HJ DíI8 \P`WwSHŇo2o+s@'=2ŢXL3z xMxGJx_6 sZHѳ뱣_g򲵗D<3`>Bo)CǪ$[\na!؝oN:ib5dO'+K.W G$gA6}֦Ai۪sx+C C2g&3>K xE;yl YqXo p'¹bcy)yoQ_&7X@iA;HT>F8Iċ|Gs.SjH[KC\K*3M8jQ rљQon+[0`HuԝJ^[; XÏtME54 @ǵpUMTS&k^YA`yH/aCXY|$z{7>iة; Mc'<q;% ~uh v(ӊ4sŸ^yyhc}/'_gxW`' i`0›Ӏ9^NΤ> /uE'$sU|w f$׽5=Q2 H/̀dÓy lVx$o?\IEzMxfv Iăӧ_pѤK5DB3[:Nj3FA&eHS5K(>9&jzJԑ#@mdA VS߯DZ /;k*Zn[L4xjg1Nw}}5eYfDP6 ")=[ƚ,-n&i+W迭BӀm\)&Kǻ²55r^q&Fy::5O4E[!4+^t^YJBOMPcf zw9ki(_H9}OW"x p$5g\ضx}%xmT1NWh7:κI\Ar)u5aF3f^b1^Dw*]u*,o#4Wn4 dj|dxv-^Ud~{Qx7kuEVmd{v+R)z L-(וn,gb2zJ_׾JhT_}Z:fGG I=>[k4")'ivIq@3;V<kP$< rXabA@[C%/ilbn|9{3`QH2#b-- Br7z&n`u p!uV6\-1I Gu?oԀ?_G?NH0`Қ W o27 $֮Iv@s4*fǿ|c=N'CY^_r\p+dfnM1P N%,Lƪ恘:<WGV#wE 9vRfC{G<*P1~ƵjrȄ Lϭ^7˦97W_T5BMͼܔBCgD+cs2g {溚1.Oi _jVd0^DuU- SdDEiw sܷy#-ƹ.sSDfO6Dp3؀NrHOMVG|5E-pɪP7? ;9h22;RԪTcĖcu3ז]xk^qssSl[ju@uG*A%Jѓe-TfJhjYsP#4 {ԧъͥx=S7ߤhif *"aT#4!XHގ/,=O/ %?# )|YUN`ig _rB` )>>RHlxI2ܭ7j#zYbrI/KX@N1W. zQafWXnqx_/L߻\^sȅAH-fN3w*@jYӺ' %T6~r^: k;pX>&:2Il5:ٖp o~G{uQXd[Px@6yYwZ<$x1$¶!88b4"A(J::.8[T~}:x攅[|5G7S$!}eLGe]ȔDZIL{zpY} i1뎞ȳx9'4넎,i0{L)܅F4⎵R$jH=Y0Hsq ƐX!UJ@~z؊X;{Q4MI)dqSӶr(8[2tEV+i-!:lyf &'N^4 .xOTO |30(#wb߂T/8#Ч4DdS+s ~cR@Rc,q c$~C92$2pI*PNkNZO?1 *U .aJ%$k}%~BDNs9Ϛmz @nؚ+ΰڧu :m i XrKud>DŽ_E͐0э\Zly/ ,{__2u~̚f[!?W+.dA:|…o÷=i>FN QJZ1\CMXfWh=5"$/ׂ<*.p'8K ]av\!~$*DLQh {p8D+D5#>_waE'nZ˲_"7=j1ǐxrd4 6u#V8lH(^J?@T[Id %f'ҘOok+ 5MPݰoJި|ZRS#Wk~*wCdCjKi'&Ѕe b e֠=Y=&?VK¬ i~._{ҫI?0$lKx,5~)x6F]A Ra' [`WwKNc/i|yS\Ydm=Z~|p=kwH(b[V_ٞC"KV,~uB4D6E3wbFE5{D|۠&zjסzթv8d$njCm|bh gYE/4Z[+At:!G=r6!Qm;;h|-Pg#NNӆ[݀ WcYUrТGjb۸-$0\1׎DZzD?܏92:ppqVq,7u7@{>4y$؈sEB;iٓP&(fBxH@xH;aKb)|Qdȥܓy[:depݽ$XΘD>[;hm(#(;'i&U@|ۅ0@ѿMuX}*ZSdӳϰ8@k!Uh]z`kBmˈQn8y}m>^? [phFG\E ?ǀU ضLZOb ;F9@q=yZOH`%9s7ǻ.'`60XN`Jk@$~ݜTz߰=D)sq7ڠ469']jx(НGnW WTۑB#FCgsʉ@UhTC~0+^fKKP?+`3pc}/xJWi;' 5G ,m-13ӕ^,e?X\2GТCn؈43\Dw.[JÀ^@jx%O.p^zxLrw6ba>R Ƣ诈 8EGk5|n\ÒѭȣW c-x6ܦHH涾˭ zRP$pu*9FhS)V›u-Bsxa^a^ Q)?9H㼳 9a]1nnڑJ!% ou֗xP6eS&Lpn=kɵ"]u"tQmiW}=4aP֢N#g\___*Tw/dGsRKq2RBlڛ˦A繟_tN!ŷAۉ ;Cف8nz F"J)Y) ϑͶF&FM\'v*btAg+d[wLpX\4&ˑ!g WFGVVWl;gsǩLKnzn3:OyA5˕bw8gA3EҊHFM+HCچjM,$܍+]-'ġC1/lyoQX ]1!f  `ק-W,18PBN qn2'ָMO9#,tVBr}y<hFILJ: ӆgN{VKcDN! |Y_.ju#r is4O0 ڙA[dBH: %RD:h+cAh7(}٥s9zV7rp/?>}f?UH g 6T 6re7EXQ.>Z*n{H|_dAbA;逸"3R<d&]ⰱljZ|aIoփ6<P3>XV٢x/o~FHHXPkPJ oLr@Aљۀ./KOw,Rbj4ig ~H=~HffYX{h[di 调c2*ƅ:8@L~PUlG+7C"rEEtËWG" pd4rJ sdzciE>t?,va4~Pr^:{ĕE$Q!'lLH7*ySmן^ ^6m@g gtㆰ 9!F@fx^]k#Ltͦ1A/ZZd=s:ֆ]9X붖+6"e~%t+TW)45W@f6g\=|fX6i~6ǿ)mҀhӷXm;5A& o&\P`\W$XsLN #%+ʁ9ϡ'u1bs9߆;} Fd-dt#= H-d.Œo76Lr,%_^lD2o! M[I:ӣYC-, 2AUo;XwO\UQ*Ghs[G eA`ֵ[FXe)f^Tʊݯ-q>e@pb KKHq^fX/[ |0^ie#8SpuRq;GIDv Cng_7PL%.N.};at6E}8$L,218BύQ+kCN_@:y_H|:~W[ЄHlOwv~lg"B~\$;ە6È ïd*\8$0F@Exr飫4/p3X{t;3u"n'HӚ~Jjڊf`Zaa5XK%I >2#ͧꙟ{Pob !;pkن+dAZEz̯0;@OG+Yv9Ϧ?,)2u߰n%ٮ)j>Y.i I} /jЍB0ev~Į'kfߎ/C:3# eϦ ntXp^lf^236ߝԑY/&5ڻ36e>pd8‡UUkǼe>ڄ9tu^*QOU> M!W w/o Vց\n .S2IĦf-H*^v0p}dZpe~m:4ud.ZsW#e" ?:6l3,{ dpsc!AX,Vn;$,cRtA rܱJmp 8u^|Ŷvggw,A3RRDs w\?l ͗μ[#rMФR*j?sydΫLPcxj8[t=Re# mgԙ|1D-m6UgM&02GGi.)wc!w0h>3&& ^ >:U\xs_({;Cn'ux9+fM s'>fo怦%0lEB:NS|3.kSY D~: tZ#QyYCr(d:%zC~\pj5/^* ZV6篵obwh~kdS \Ñ֑{'An %}C[t5G= > B5dJ7w70hgLvXW,4j`ȝ] m;*Z$TF, 'gj]:{-Ob0޲[^iU28wv?`sꖞ*)c_s}& ѣo"]*Ls'k}8kXo<`w~/8UP`-^\?AfrzvޫPLXմ8тI:<~j W㊬F?ϗ78(/R<9=ofl}go|:%/:ES|bJ ]X?j,K4M*/q^ 5Wi/fFX$Q0uRw#P/Gnm"z͛*aopOiFS5{X& lԏ: ^q'8|$$ۨ!rPU W{7I]%<͛~ӭh~xCQR4Ɨ}4}^ 齞+A5Z b=.v&&yGk%6%Úas;`t9#t%4YDM,J_&c $> owTbF'ڥu-v2(&NOk1LĆ'[g:4@1.vrT9WŷW`^Jf]Nk(;wMG(Ă_S.N'O? AA6G vV;IF=`^T/e(! B1ҝS.^-o9& iw\I7cv;5$9夕*3M;xN`Gۚ֟$)q2Bnާ2R$W܇=?@]ns4]^Ø: ETi2m۷%q,{ |VS?+>9U¥} e/JIi*0_(U昴2H)DG+ *OjN6$XݭIm&GPˣiFh|HO0zH G@_|n6wBi j8.^Z\!?LykED ս:4 BuZQp,v|abq䨽p':@H7_TDqlY"jbBddȲo`ܟ:P^:4G+=1'I%=ݾ3`Ͷ{qz{Uﮯ"ت^asSVQ,8ZR㟳1b[/Q`ԡtp`no ThPYS+Dj&*rO ro3mBN*guR?"4{dC!jdׇ UXDzܝwcIt bfR_4htHn'97bv-@wTpBWŝ&rg \0쀸UT]A)YVSsoQG6\\250;SKC2'Ϣ$n pXF2r&قA0hXQXt*C/'+X~ZT\b]?OGΦY i6P>o, ɜ|ҹDOD>qlm3¹yM\44 uu * 4U"c#-ݗ*VR0ԴB ZBw(16NlmkH#h#QҌ a zm,UA}-t1BLk2όαL}lF?<9{wIAt:K#RV$$/މW_Tc4IPX&-۠sf)ׅV8@~~Eo_p6t+hKUb O%`Qك]D{xx "LC{#vj8593L$ h&9F+Ki9uS@ai.$MOT۸{V|R!ׄZ"wU?f*t-Kn%_)vk'Ċ "Ur`^>D4mez_\22VCSH_>ijSu}<+O,=#|hB937-g.R2B9_Z~; Y02U͖Jg>E5\Վ@@Y8szK#KT^.y>RbWŚ&orA#k9b>iB =E~v lY՜B ha9u|C.9s3cp`Z7o\sL%y9ZsΝ\eV9|)y YA"W"l_Ĥ_nhHLee wuj8z1K4gLbeM8"]CMHI}z TGg)3 ,^-= 1OP$? nB*Gu枰GԸ{0V,s0i9[QrZM|Zo&4F"ƻbnjĸVŘӨRɓv+KZ֨L#U{?{e4&e䦩(T%d*8Z3eB y8Au(W{NSevܹMF YQ= vNӢg+,-_Bl籰+)zm.eB <2#]:lpPAۜf o$]ְ {& :^0Gel\;a +0{0k6`ʀ:qlpr=0c$)|6/uyԾR.Dͬ9@0!< \S4VXqDrʁ=UƮ8Ce?~ 2D9Pj$ fe3TwNKpU(7Rp}kB1_ߘ[<sL,f2J;IA@ o&Vs@wHHEUiqkD3\9_<* ɋHKCSLIeycxo#K^j\T@oF CdgPI-DṢН2v@oLN03&)[#gs2펐]^PU/L*5{DPjKh-JTh9gW:WU\6Q,z Vgj;Agb1z%>+%GGs]0nRsNxhs:V:ܿ Q_*7U2~$\+|J~ßh`o󇥹J >ھEeܭa' ڥ!^6W  A:yTAEcH8M3+8oM0Rкo9%s=+!n ?x "ѭŴ0x̐D|*w:5## wEoIZ%j_xfղ@8x?RoOՉ[c6D!6OÓsh 3.E+i a߳io,'v =R,NSv]4sܣP)i3q5OkDV0 `;xspM-=/f$j]s)i(ꓣI j7,? o4 .Cut9Τ"gkqKg9 (| ep7QQV.qQ%\2=1"%LҔUzT+D%QZz 3{%IjhyXg y Um]ixfї,Ybl"R=y$jf/,`P(hY+_=0?qIaqeQEڟ\^Pl{Z7S+5\t'+Ue$pE@Li= 1H4xZ˺]I^ 1t/o[Z*чz^GlȂ48$i|$щQm76%",])N>R2tRCyFIqAC:K01S><˜bSZBDi^e, 93E|R*.UL];w`M>(Ƴ!^@Pj1Vpzm>Xdbf`gh ݧ({BJʗ%QccG:VEѰϸ+ឫO1R;TN#;U]8籿b-3tǃ3f#37 @*7D? TB29xKeu]W-?(Ĩrԏ;&Z\w\ ƼdfH]6ԡv7=0:恴lE}3<#QFM8t~Qۨ:1j 탇*ܜך1>D__LKIܧu͆D)&@Ɠ:@ n߾Ow"RDG3alrnN8{/QoA7 ~w BOךQ;Ҟ]OZ%hx>7`Cs+6C<~rTt4#J@ӅݦER#5xoF0ZsAze./2d_R҉\c$# EMGnpޫAjX̰&QIbc Q90lfxhq~*m |gG6_1 ԋ#hnCy>*xf o&6 KJ<Ȟk:a 6ZhZ wgm[ i @=_U׍&l 2bԸKI!|#QuKy-=6t۪SYOPZO-ހ"F/A=K㉅x aG0Sc1ɕM#<lgA'[?9!2ԩe ?4D5mr —(oU^ !S/hU;fΓ~DwS1 b֪hDQw1}aљwX<铊TO FE*ʹK'Щo$*}Pz^<fax9CBӞ7?t[!@]Uw(ƴ!T`1̯? ]HMn nOZxJ3?A˞&{2_& ~ n)(_y,Σe1O+0+QdJߟ5npΘdC7P|}zmH?^%3Dj~E?\t&ZE|ː2Sh1rttN{OZp{)\Ss`qEg8]ŭ)IH]zeR!L<:&Yڜrm+՞:K8t?;B a?5<}y1ҷ{vi+͂$;q-x*UkYK22[DSOĆϨB!|J&\W<##'Xg ikx><%3|EpB0 ݊>NⱥFR*LE=txQ5篋~W8^ӑq~ [1n7|qjFh;BThnE_V[8B %7LW6Dn簕hKFc(IJȖ8TEHSA;BcT# x/rp(U>.tHA+lU"=GE\ h$X-f2Z{PqsOǕ!
KC}&o[-l~31NzَEy9\TϽKDȚ E_&|璖{v3Ҽ؃fbzp[+]uTQcz0lhxD>W2&,`z_%9Y1ٓCZ|R+A9^싗O~|Zw1S(n5AU,vuZAEMLC"WLi/LϏxG%a_Qj>BY89b%w:8'‚We E0| tυ-@Xe6n$s̼%1dMg|>9p VkTye7ȵ&/ .`ȚXB~-V:eja/?>[}h$}WZ7 h8f+1g])PLT_,Emtzi#%xUqQ!B1-#yp?mrNʹȜ:FR͆ɐU㙘Z4|!+Q6Q{7Z^׊ݑ@oy8)P"Р<x|8΋5$5wɦARQ@ϒS,D8KlxraVL$60Xȿ&\ՏNCt_6kIExFkKEvןʫ&@2<j\}EtF:#P[+τ5B|=]fspt%Ө9uϞ/*nXtPؒ>ӺmC?XF*\3CTaYS*eD>Ъəxս-OJ R;:T.M僎2Biկwʥ;54va}0d"o9U Fܓx&ҤZ\:p8`|fQD{tZ߄Zfw&۞܅CHRCI \;_Լ=oW"͒c2C*>εE>>ou_.>Tp7M4[V Oynh 2:a[bgЩ=ζ YZ