asterisk-mobile-debuginfo-18.12.1-1.el8.2 > 6 6_6 3!y덏%!E/֡c !E/֡ӷR7avl~c/vd*y;R7{POAcMje{@rfV0ST'}]h^eAO}Skv8ED:N*b::lOy%C|NUkjiT3;PWb`஧lxCNg8OgD/$`Ǫsg籔ˤz0JPCS\uEatd[Vnǂyb9少y!fu㦇8>%0km."{ oT2:~!"A>J[9WsGq$Va! csilE;fO㔠Glh]WMDxTL2u=mPy:w::v178Zel4Vܾ(oEꋬFSHj\IOݓb2 H')& eMy@:94ef85f70ba50b608f69ad491996633d5c5a941c0e9c753e8276c0eea9fb69263d4c0ec596e00a6faf5e2eef32e2d09c1636bcfat3!y덏%!E/֡c !E/֡-'89*w;B}p??" * X 8<KZ` o         ]   , Y  0(89 4:Z-G H< I` XlYx\ ] ^ٓbFdeflt u v@w x y8<@h݅ݘݜݢasterisk-mobile-debuginfo18.12.11.el8.2Debug information for package asterisk-mobileThis package provides debug information for package asterisk-mobile. Debug information is useful when developing applications that use this package or when debugging this package.cbuildvm-x86-20.iad2.fedoraproject.orgFedora ProjectFedora ProjectGPLv2Fedora ProjectDevelopment/Debughttp://www.asterisk.org/linuxx86_64<cAAAAAA큤ccccc㴨c㴮c㴙c㴮c㴭6b91a6efc70b52dfabe6b7d94fadfd1f9a841f1099083a720702d1cc88cd8fcb../../../.build-id/67/114e77b660273495fb2742d4f7eea373413302../../../../../usr/lib/debug/usr/lib64/asterisk/modules/chan_mobile.so-18.12.1-1.el8.2.x86_64.debugrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootasterisk-18.12.1-1.el8.2.src.rpmasterisk-mobile-debuginfoasterisk-mobile-debuginfo(x86-64)debuginfo(build-id)    asterisk-debuginfo(x86-64)rpmlib(CompressedFileNames)rpmlib(FileDigests)rpmlib(PayloadFilesHavePrefix)rpmlib(PayloadIsXz)18.12.1-1.el8.23.0.4-14.6.0-14.0-15.2-14.14.3cױ@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.el8.267114e77b660273495fb2742d4f7eea373413302debug.build-id114e77b660273495fb2742d4f7eea373413302114e77b660273495fb2742d4f7eea373413302.debugusrlib64asteriskmoduleschan_mobile.so-18.12.1-1.el8.2.x86_64.debug/usr/lib//usr/lib/debug//usr/lib/debug/.build-id/67//usr/lib/debug/usr//usr/lib/debug/usr/lib64//usr/lib/debug/usr/lib64/asterisk//usr/lib/debug/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-gnudirectoryELF 64-bit LSB shared object, x86-64, version 1 (SYSV), dynamically linked, BuildID[sha1]=67114e77b660273495fb2742d4f7eea373413302, with debug_info, not strippedPhttps://bugz.fedoraproject.org/asteriskasterisk-debugsource(x86-64)18.12.1-1.el8.2utf-8f5bdb672d10921ee5c2758377ce6b0e970908ee342e06afe1b5793c3e59ba2ad?7zXZ !#,P] b2u jӫ`(y.r#޳O%k򷚘jjɸڡ$H;߶d$<}RD³ /c\)/rWv:w՘ƙ\1b[!-'K!q2_nf?tהuغjt 5n[0w`<`:f񿥭Ѿ%2f;ch!;ض਑ .7u- & R$K`]?o{jDD7"{髌[8IR$s]eR-̊޼x::r@@jv, ^j}HʂR7}|odR,A^Y<'wb'O&;[P^ ?XOrTf*ƧpywgH 1>9+7&ۍy'\)Py=6 ! R AA'Q5Kt?;>jՠo⡫TjJ_ɑS1 f3tm#7r/@T5aPSDxjț4`(z?MUveQ y_c>}t孻(c_yA'[pn$a]r9 H<9V5h=uBYQ ZTw<5 p|g3"==pL1#$1N1pO0y'ŭWi8iaYYWQT>.BҾ[Ig*{yŸJ.C%ȖrdGb%Uf o>zIۭQXGPjT4!Ra)툯\6bp;vn(ͪDᐸ6fA.|ŰU9\Ē-~>Q:PѪ]ssyqrqI)l0줩m,n٦d8)~[;m%yGsc0&>LڿeB[d8x7{<-ҷp۔dm(VR *$׆ci 6Vx#JoRjlIW3]]ȏu$ܸWX6K=胪9X)er.mʪόN5SMqޮmqKH"|>m\5U+Xl7H |a$j\ b~QaP~H>HyC9ԘUAqVjAg_*z˨|EuUFYDzcc >0pXA{B0 {,dИjҽ(k 'K+Fw)cŏX[OKGk=oCB/٢u.(M3;\Pq䌄`'#tO0TǾ9/EwDbA3HH!_*pS}k0Nڙ)s(D(+ǟR -{LJ*29m!1}ϸ۷#tR_q_D<ܟ™:eɱi4gZjˢ[.7cE+]'䁚,c>dm ,|4cmXA++9pkpQWQ: ?ԹTݪهKm̤>TB)!yZ"q!uxF;0n0!aq:Y1#b9g.@UTp-mV[^{h @5IRT"u,[ģDPmc<-oEEoiIM= 7ni-hH>Ek/m.#/Xs^=}aL:FhNvxn$]Oъ93u)&et)VHX-~P> o:n5&BxsܐQml JUt_R~̹[D>H@f ڂˉ]ٲi(Js7x~ETuWS2azb0U{r/"?G26zSR HD>!`ȝ! :`;.6W7_/Q1'GךӬ;m_q!l*~ɾ2,C3GaJE љ)V﷩ 6\fmF̀0ͳgR?* %n㬗znzں!Fn*wX?T)4 ĿT $Yy-^㏔"O: D`v'Jǀ Ʃ P|UZ2,n̞qȨZ*0!ݥ %Uqkˆ՞.k\Yz&>eڪN .D0  Z#L~o1ٺnX< fA(wHH>Ylo5^60 ndfMA+X#A+拼!nNd8QUOGntZ=O@{^-IS7-As6Lp@CNQ+h$|YuqSy2$JIJ8!} y"i"?hؗ+]PJTJ&,*[\NDSӪKc/եvp8%2nGuV'+Ԅی_SVr l PGwyd4xnI,$Q[YU hM}DeYJJt|%fp xpl PυR=,SXZj'!2Kp@5rÀ[S -qwV }^}"d/50Ej]2V̠+"l<5Pw2_gS>L4srG= .MYet"z˒%x蠥:6f5M(S68b7)6Ӥg]@2o0Bj6mpE}JllYUɉmmXCJ8ꊩ}@k*%P/ XSh#MxKtN^UDeR^]#F"NIbٲ 7W"]m{jE^6[EmPU(뵒-0PhChwI}%I?J$%bol"*vY fL'o4)q^o cEDY BЙvL& HmAk"d{>Xc۽ۣC`\ɓg9 5[[[!̅M"d$4 IV7" !ݙSVV9l 8KG5ɰ Zt.T.K-~V;,ЀwK5SQpS{C@#ȕjY?& ʘѼ"3:2MF+=2S,+$"욞Ga4irQт ]T9сT'&DҢlPp ,Znt܄礗!U{V`EdӈNDYȞD%= f1rb:6:ӶI+cgV ? acVv꟧g2w Tx*ex0ϓ+W{GmV[ O!>7:F IpVۡQS-hVFoMבYhM>CɕS_*s}/Ѹ{hvZEVv 5?n'[[wϹrՙq_1Y/b"< 1Y#bG -"{ӱv5wU/4C[+Y%;rXXAUOcsq\js)y͙W.9}Wo0^6N5Xi 鼔x:k5T@m#4r:SO[å)Ks]| ]hFNj΍T@L™_6fg^)%eCkފLC$? YHkV!I -C=Q-0aBoD\vb2KxTM/EA1Fj>˭hܼ PsC@Ednn(&}x*g`p5xi cW?ywz kRhv"uBKϰ+ş=HV[@VW* Oh[sEq Ì:k(t)0_yox|"E$KEv ت} =adRyX2m+ -͐`&}WaAK}?ZӒ*x߄s\'Ky54M8.M#<(M{<.3YaB @)ʟSHwcv'F&C܍^1 tpt @ n=mcn;}1?$ sAE;xUbMɝ:Kzzv29aϾArB*lGjC6ϠJ_i!v_EJk~v٢<ڦ֋#T)W-_#:0eÛCM]{\{j\ |@ȷ~Z?j+2Բ_֠"(d\XopO~ZĞ :ȽɘP^)X[X;3j śrsiQeAhExsx8D^_e#n:?hlOAKw\+-gc$(٦ OMGfV!X}[YV{<#72h;f^C+gxRnSYRzꊆS;-,> Q 5R!XK&~)fY wM:yBkObz?X/yg؍R{ȣ0p};H{ Iz ozmsy/^ hJՉьuYkciI+ U*oT1jlQ#Zn/Grw)cJ#f5`]`Rb$OkhP!\^QIo5X ^0 Ƨt~pf1ŝmb}at!>kE%Qaאh얊iiCq`7CȫM~:h2RW@߯(SQr'[/Reƞ@Ȑ>aBاаԉd_Q!3Lpr΢k,H e'Eʾ 䰌UC6)o؋Yr[sֶ rw/I7U4D2ʆi5@.S~R FnsmtP4\JAИzvxz֍rJh婦h;mhfw~Gs$EI>@r{ltFx<0Sd:wr~Ix[29n.ô6ZZfATaNs_PNm ({:D^|v?]o L%EEZ;QS^oqs+s(4hǹ6{'*-3_4֪g;B ]R$.>X+ݒU1BϾyN}A8TD<qp )%LA6O"0mŊlRIب$97qOwLv-7R0W#qa4nK}®zر&FȨI}eH>NG%k_e{_6=ĦjnW`j̕]YϣdD4Xگ ]ߨOw_KQ RmIi SL YFڸ~Md2bʛkXbQgoT=`d|{r'Oˁ6Ñ{OXEG&frAOzH\I=z #ZO81Yb;[Ёٿ6qkE89bichǨ^x[ sYmۈ'==y_S`Z&#7>RWR؟PM!_y}q Z;Á_(qGs"ŽHp e"_xa胩/GI)Bw`|)cMlc^4W@X+w8n"JO,!M}9-N:r{ϵ$ q]%`ia2g}wŭ^9K}5`!O?Bo0>.Rop t7'p0*HO΢*2Uf[VE6.PKpJ ~ NA3hz_Hɛ?5[ڣBK(Qvx4R vr'f,/M* }O\#j*Ϡ2 1 bQO\3E~g:yBŀZڣM+Uz]]T1 1i4cz'uP;}NE!l܂@9+$1RdByV5l*V|l{sA\4E=Vqjs1+=f5y^nsM GzSL38 mk.&[K NNay&Iq:.77%XKmY+>?;>oF2!DTKCQkhlYznO[$w>٦O/[RU+l=}i3eo"Q %8n`F=M#ȋۛCC۟~); 0p-2 i&WF2/Ank?33}e| CYҰɆʷ?HKјr?l8kz/j8QA S"@g,rB}g|5Fה׈LX^Y ߼Z*I1}|2R.F3}#^2%ߩ )Zcyi Bλgm*C4Wˊk[~v9GOHρҀ`oTG`qqF)S݀k`%U{LsԳ&0R,- ҙsӑ^H <7)̐=*IF`Zc? ef=MG n1*4>W.ZdB"()!Nѐ;%vVU^KiG/a":i> k Aa䗁ϭBqIf+3AS t/,Ž0t"1 eʵ>Pd33Q"C&u ٜ5.DLi螆VDNS erm}NOKD9 !ć)n3o10求]j0l>ydR*~&xBms^gcc H-QmD2xy|.ux,yLl+@lRN Rޚnfe더+jãOFT.kڤ<(2uel#3<˲aw"T}`4XN*.~]{ p v_A Cqb,Y9}.GeS=|Ϯ m݂K*V3.n#?mSTwZ+hTx.WtY}"&bA2xX99ڭx7lf~-8Gkqw64_a{ܯ.LsEvL5UUj?Ea"4i x]1Iq\S-oięuoffi'Qy_Z?${۸7 k`)pCVMZ|${U;l ,WxcJe B; oH.'`ح ͎ϸ'_,ʓaY`RX~ R](O(I* $xq˫Ãxr-dD9A|V&AzѯKcBNYK[ԕnO ?Ҫy[ 6RcaRo$k=QB_f32-lJU@"Nh C2f~L'0D&Ho8s;Qwd= JkkgXh9A_! #%ʀxX-N$u?A U;uLPZ5D Ki6l}m%69Աy#-AS哧Lg\"7$A߰93@ϑ` XFɶV0$0 &>oѤއ? !ˎ͍Dz!{|'( *%>a t]I'j c߅xyNlv3X@+xj B\^#xhpqNb C ER鼡Ov{ K57u IWIC=K8R_ YNSjj )W2{BΚ>;'%r5)ww: ^R^" {#ߊӁߐułI_I 2%WW7 Qn"RJ+Rη'gE-hPYG |پmÒmiP3M"?rYmx(sa%;lLTQ&ź(_2}R,$$X>hF=P:%Xjdd*YՇKް\xٌ#xH4ӏz*~ w~U[~h*eJ}N!튽䵓FoC\ Ѽ/i',~z;X'pץJRU8FvY(o%\HתOEvFCM_ h%ej]?Cu+ziaj5h\r #.2cy-7r2TAAg/K >QҞֈM&aBj.PCu/1{۶do$KH*|8ɒ@.|< P9;Aaxt?[/'%5i?Յ7Xk,gXWOnuİ%Ūa㑭A5#97, íE<)#rӅ夺2``ðg{Um1[7:-̼]<E4WmT$ROoy1,әDRp= ~ᨄ9ap,Nٵ]WaxzѸ}RkXuvk~I)`޻|eg.aӓY(=Ͻ JZSu=W70\lf;ȟ@pE ʪ[wM_х^^}=y `Pd&X`w,~/BKw(?qݻ/pF%q?b:[X2N2:"U)`brqB"+MWc'0{DrH1> =1n6*Uڢh֓B7 -&@68\9=S6IY,v1[XP_ef1%|DW!>ad DQ!2݄v6b $_ʏh~ywv p* X>Gn=Lİd* &V ?:`|.Nb<UuZZi{nY@06|OukK;M$ l j~mTtSF 7dskME@ldu]y A_@c/V:XĂ8Y?uHtm$E2))"}ZX;pc  T3/3!ˋSWl#hn4- -s\T&]dXYpUk9zytE_Fzu~K|Mb~A1UL8(Sر]0bd |{U-ô6Z'+WSiУY^ O&k1t~Vھ̹x{r Ȭ9JA 8\t^ŕokꜗ@sr=¥Rg@gMؙKS2|=X= Y,ɴMd4;5BoZ bee!SJ^zSrԻڪFEa?5#it,#c9 S(lAq|KR"q l잽)-b3*V i-G] qܰSFˢlug,.&:0^_tp/<"q1EY35<IbU"t}qI7oD60=_"}53(k}-bх'u0g1a0}UG*tx7ў:/P4q3!ܔ2FLQ&&B'-`NgL:3t8 .]vM#$Ec܋5nz$\U yҏ+!c & i3%_%B\loR(ǫ'UOx!ɟ54h?%d-(0=2hбt@䴡0ޮۇϢ M9»OI~w+ȑVԸ}]Nۓ]ӗ/LFa*!e☻(螤HS~kBi[6 )ױtCP {24A%eNym Dߛ<~tz} c~_l3a9ȑro XN~V0?_ UE&C#eGÅʞNB76 x 0v}Ni$uꉳ<=YM]|P|L@Ӱ,YW]hJ`nͽm ʛ4(\X6)%H(r U:9>~HC4{2MgEP\C.=dkcCp5dIuBjJ~Ԫ%ob:,A'Ko5ٙl zso0-\-7fw> %ؾ\a/Q\v Bi*S/ה)=oYcTBJN\-Dae[ ,.l=:A9dUB碳TgZ_fcӃ݁u+E_$9Z8 7nVa/_^?c_&4)zY41WzDeB2P٭1^"ܫz_iy|˾8wUrbv#p(/W0t3hzvi{kpW-Xk_.(/5M'B`OO$3L+l}5 TE):-#p[^@T9wc -3.\a" (}sTOCsm T -16_[{sP](5l qAo~Xy/q=m2 \wꭹ'I#Us7>3^SOla=YH&J“=œFEt TB.K 14iH(~Eir3):r*e9/`gk=:@1U,/KN"^~иU9W<­YF@-ʆ|FKE|W&SpjY$o_0bF2Kt=d|c=%,e81_,d ZAʍxru OAaXkB knvŇpw@ ZW"?B%=[f u6hW#͐>d|h Æo͛QCƤhRt)ـg⋚Ug$,v v0^H! x-=JvG[y]|šq,[.^ylkET#Xٜߐi/C"P!bxlő[x8tArgTR ֭TzGA~QL/;'pr?KS34u,pi!pczcW~@i1lu&~h`yڒkfmai q gq/gsjGzH )#-zyB{C{Q/?RNQs?k!I)Uxc{kYfV!S¢O*JynGpw?b{Hh@/ԙJq2 bE:r)SGqo V~cUf@{.{e2{VGl<#YvwZp?|u౼'amEݕz?';$vsU\I(npG\ُ,BV2mv qVItSQ?TZ^Ŝf խs0y)ո @Se;2ry[ a ;Iy[" -bjf+Ӝ0dTTHsV"YCƿVxFd˽Ǥab9>NE+(?W.ʹ F *^iZdW#IԸQks'w"'h'Ylф\'B'KY P0dMT,&HKnF2> L3*%jJJ$pSvp33p+ D9^^|{h:bSa/4͘K!s Tζ-r$iV\*=1: d, CsbJWǓ苐[h! 7ݴ]ܘixRcS u] ] w[7i(6(vٺ}Ǩ"jZ`kj Wh3BC*N΄@h+X# ٟdSC61YcՖsKJ  7;QSz&Qmћ CAO8,ɚ%Hbj>WγQAQVyinn{wUP}Sϙsp I9E3-Jz,F c'|CO!C k6yHvn9bekFz+a<*ϑ}+, mۿApjIV<(z50sARYŌa鶪'*{; 3">bHL\y+2v#-`ji;iS`z̍P2nh&R{=f:pàWᢠt2h5U NW_@`\iH#1Hݫ! 8J!]mckQ RH<{g%ZCa p}+UѠd< y|" |c DgR-P/k3K};O}z lw^Amܜ e.%hY.uw7w3(+P8阞ȥ=Uq#<@艮>:*4[)V\LhcSZMr ZËLZ%eŲEEqZ4OG_Řd3H& k/E4O!)sjpU>0N/Hk5;(|D{{F`ڌI7@ QvddX^**w@׊56. VZVdS΁kv!q̚ N% *,,9}VQ G=aB9fc0@FWX,9J XrZ5w<,F$י ImOl隰hP[ >ԏ@CdY^HS*a{JAh|i8gwE3;J7#$hGuXCZ(Jz-w~FF]mԟlJP51h\SVG4fGo&4Db3v ܚP''e[xp<مVˎ{z@rHhQaښTP[Bkf)Siw7ZO ȢD7_GXہ\c߹X(u9$H/aY³oUw%b3?fϜf"pq<59p6<⩓?|Ov8{7G(A|wDgD]sY>j0.LՐR~6HSn1\%ϴm#h"~[-ȀP؝#dxfӀC=~ԡ;&9 85u#娛8{3xx7)9+qN9+mXldK@sN4;[ 3{*%.؀tClmYB\A^Ě͏viWnJ'x+N%ձtvU;m?4JDmeWц&p~euufA\W/ĩ@,6%SeyUTl6{FFm'c5|yLf2 o\S.!PH6ڦõ/P*v3Ēez@,_pp2MUKnŕO?P>] C҉w Ȭl;{ŵd᭬6'&aAYLciœJA㒉d+ODHTh3Jfo8zE6ƍROpgxu苄ți!YX]?-ɇƎ?V1 zW11,}bW(@Sxk4= s#R9&]*:q"[eb F$pEd0.$}^7b*~ `ĕM6?d9~3l#ʝ]I#Mx8=I$H ܀)狘75@M^ cT~OIwC'B5Dʁy#Mo۳N30]i/ohl6р /2mb8}Y YХ,@f T 3k"ݫaXgo惫2{@Md}1_ o:=Õ>T3&#F1V- 1v-LH>ʾzU :+}l#q%&al6ʗVt*u;7hΊ$N rx% lrs1To[ägY ӛ{KI"٬ dro-a{_/-3+ϤAYs*1扏6'Ømvv~)ܞJT&T|+,W#C̖6$5%Ut؊Kg!wљ heΧ=InS~CA#\PJWܕƗ=ѵ8>[Z n{o' 94ޟ\r[U{b_xdy3.$ &(g(F矾 4"hl,淨ګZuJo߫wG7Z - 5h>znsF">"d6~/|i(E@,U|tS6.NI6$z8nZ!iΈRUuQEb9!I?T9y :9qGIyz c+QzJ$0"CMPٔViT5.a#R΢m,2F+v}13;W{1/2|N-|!Y=y;!8=2घ64. |hca] ^2^EzD)ۨ01;yfS  M)N8?+C,eH J?wu![ =YFܓ {ݒ̮dvڞ[Gb7hƞ?>  fLCKAmy@| *+3]P=0; |KZюOsazFD0OaJsvr`f?ևcCM$cۓ*Ps)(1 KV)U6و|z_e3s60˛ s=0nk>S4ˢ+${`􃻺[Y!YS_Qpzbnx*˘Ʀ_ q@\ C54W_쇽a$C?@B:5$oUɬ:a`eIbޟ=IgǣO Mn Jud|D3Y2ZA)9V^]gFL.rlQfJ-c93wx\ 3zMo H5a;qb3v RWcokLFrg^_ Im1g]U9?=e,|\ 51변8 j&gXq\=S K2v(vZxi14My{ IFpPնLw-iVYpvKd1kڅqp̆vxwSL}̴fy7EReɥ\Xs4` _.B!MYFw4/ ߶4КβRL0csZgijg%W ,3{`2-xU]ce!En܉@1+GYfQcq%>bZ+@ߍq0Ѓ7as L뫽+QH!"ʴr"*ՒϤ48 M~KΩf/Mf/m>xFcJ$0uAy+$2)r\%uPfKmM`^Cv#RT mYngqۊ>BdAeL^tKwEҲGR-5 _ŎvDU_،P X{cFw"EbD4z4 Dr|C+~m1kY⭖pT>TJ/+c~.۞`mGR}+R"l`2^aB }GݹY4dӥ9Z ҢX z[i`Fó}K4byeûQԎ K8z^S.$Cs4%` auKȓ="_ H{c듉F1  # (vuע؂=U% טni~7ٚ5;,TR20HE浥W;dٯDnAhL ?My%$.hx]-?Ys!ЁBnt$ xn`v+ٝ%z=&Kzc2\|t@@bk:Yk/j [A|U'!߼Dקo[2|t(2Ǚ6V60 *fU cF7SBSbc'TX+#{$J]>1I`Kcيp3ivk5q:BGj#b&E5޸L|^7z Y^ 4AqXd?g@_F[6-Δ>|̓MQ݄YD?* }yVi:vo" ;,iZ/RD2&'bW0D*3Q_G+|p V) 7ؘI=>ONl៝ZF2d= Cv߼[|.%>L'0Ҽܕv@yG{S[RHW<v|T>8pcڲĜ*#fX! PxΧW&yi Ղy5ң }+0fBZXB[<XEraR-.g#cz&|6ƛ}S9 A.DRs@U2٤Cލ M)Pȶ)_K'pSǃ7WʹѬ3i Lˢ7 F%EDv[J궿vImJ5x G;xZ@v"Rp S!q>Gp"bd-=9@ilnt1CS  ^m:Uk\\PS?kR>y nH:bW!وs+X0N8K2NZ"rܳ*_`7=;/~_)r\w>o(wؔAAh,@"^;5zʮJ$l^m/(^\v%vt1$M4k|Kv-]b$/b^lTU|3Xr,  W!D8|pAL*N]1v0ɇ;< h;^B"V4ƹ>ՌhZI ntynǀ # ZiZr*X}Thyyj^TC2q';Jd0gPY.A?qQ![[o쓍=dLs9(eXJbB98yhLxEf$㬠eHk:]\?y~ḄXSe:``)=Ha5IZAr{^䆟I;9#^k i4vѓC8԰l=UDجԘZĂ oER߬hKc`ff|K;1OSCw0O +*=V"9::<|*jмc9w~Ro٠kFqGIҵǢ^5q!l%˟ڱfXyb >zNSTh|1)@_ra?+Kg@S%%$eAnkŗD9_jbз>ĪrTXfR~-H B59҄v68>Ydi: 9t)v;Z@J-f|i?Di@۔qm {BE!p+WoXEN@9>J(2\H0vI]膾*Te<#Zģ8%W"<v˷K&qGzGc繢 dt^E1!qffĔyHSwEZZyǤ)Xe x+{JRөB<%'BXzCE,+`x;=B|5s>mJ̼G¿]DF7sEyo;IvD&tުhͪ.DcrlZO_yhB덓se?|҆n}ƒ/mv{c no'TAy &:rѴA<c^2.A0`!([l^k;DE!(h1m$ _>KZowSn)h2}.Im!& yl+fIc0=I j` ]JF^T/MC馌!'$tSzJ;N{g,ZҘT8碢W)^b'0ly Z:`U;%b 1 :hVbULtByDJIУT BIٽwC[{C~4^އuȃQdo ~[ `M].J|/-:rT$]P^*W{4+ײ%˶Y!/F|>z`rhc<,cd#挣x >~ZN̆ƹ:/9x.cH2ys8V<[Wyl5֟i{Nҟg^p++HqD)ǚRKTG!etdGm0# .ÒyC$ZtZO#]m7F;Lb8X5ǭz 3F^:J_:qoqp]#)A @(-o000~A+ NҳO;{5|N bs^&0@{XoXol} \S|<*,r9`t$$GGq(1E}'=ϊ ݐU SKx04${!G][; }86c*c:}B/% B'xK*5?x! wzJX%wa)+Ut}K ^CϽYEۮk`8 ,miM@x& p9tvXR^aYбh}f wYhc✾V|9 ;شoctaIw#I:VlRJ2 o rSʋ|K/RO 67g M2I_c?Ķь'KPONYq޳ָRJ(M , 4Fr˞`OY(,N˄D]UZdnTvOJbM @Bw4boΎbrc-_s׸jaCɀy$R)5rMFM".9J7^.`Jo~ɥNW:Hi Zïԡ'+k9OLeӊD >ol^-򨥏~ QNyAdƝkCO1cYKx,4P4"=l^K8y8hm"yxpaϥWRU"GX|GC6^P,'0?V-z\-=vU** U"[z!A^XY< @R8_qZ',(Ӊ~Y=ۭ BqM:7NXkVӚA`KMx-F:"1jƍ ?wL k'hR<8&o UUٻဒđO-=UIXVF7yom[tH*̩f'~b#^Sݹ*$<=|85I13ksR\r!^_@ء D<3.2Xg@(6bOkÌ`Z/AvY+Uyy u1nЦ̾pOb>,ICF/PAK,fsg)h.dȲ*AE< t xo:^E3ϣ*POMVb oz4݋2=ĆV.G,g%h=jk:c f^Л` Aɑ܏0Q iS[Xr}pgoZzWMi3c|*<p_.WO0;}b){;Y{}Dl*~A&D (TõsP8JII)rFHckuԿc0YlgQ)Ko wQwG^>>Oj3H6pItQ%΋Wܬ^z2_ʨ{yWVQЄb3᭹IG.qB5_gҠ5=5 \?1PƌR!wT fNCь:%i-!} lչ!Q`-)cDnAh^m+9^l-<מ3pa0ٔu/zz7 S8Wê5@_"y&9+tRR<>Gy㌟XېFxڿ$xq^:>N`~F|ڂ- -?w*,&q=x#W;!_4X+vF&HgD|8$e!(/ܵU]ktOAf]*I]Ā u5B6էIddzTB[m$f6FG݃Zg-#`Sk >:)GRnxb}=q^,ݫyC SqHpk[D061dqd XB8mtKd~/4ՕT@܇g"at¶U?S2LwAڷL QY6wUlDF`밦eGP ^9R?B6#Q@z~y 1p^!rά -ȟX{\Kň'p) O:&x$h|-&:w?+%>RٴeBC.2%F-kagr]; oFmT0b1ûwBFPxsb;\ֆsGҕ>zue4{<@(g7km0zt$./0C]N&ktJbTp*aĖMxV;%+ej̸{oWDl~labᙄ n>楽Fvr_W=GCXS 6)'iXҬ!$DChxDDkNSeYMxS,1񽧄G K(i tFjbk0@S](D-忛$|~ax$Sl|\zS ((cBATs:oǁnЎWQ\Y4[`$'d-nl7oOOe+#@qG7@N_@7_Һ[ piHK>IOizwfjݟb2I|Oۏ 0R9zW,M$-RLUahq]q򨳍Ҹw Dd'Γ \T,6=R)&`[h-d+|;h[Ny,˳,cNÔaDD .|W0˰b)L#"XB~vxP{V|z*X ג5<ݫJt/λ-)+}I_Hh/kB:.=g;%H;z|5N64D:99*%(] ѦȊoX[02ۀBV9QZU8܇'?՘ـCܕۥh"VSl+< a4/U8F|ƶVwbĨt\U)|@K}g|P?K("$ gC»au7X bgT+MbEHZܤ0sFyYc66TX*ԚZ8UNPa+G5.ҔwMic_ կno*~TCH|ewQq 498Ì]16uqWm~t&xahsLϷH4ᙦד9*op2w?NWrDOR Sz)wG.ګė:ih֠(\d48.n ,؜=o iy & X~`%)$fZ cl!&%By^[ WRˆ"\{'@!o=:8 1~>\TWav&ZF,\¦+x(~aȽ4Bo!扴 [qR5̵B c hB5,Zw(RR).3ozuwdɋ+V(M}}{k0W&}jMBҡ2c8 ڸ9Nf6 nʓ ]ӣ qR@{/d+'2؋P0kNm=ՇM?˧C8OcP2p a\+ DYQH}t˯ GfҚ@#NE4U_SύͪxzOJvEL|0h@߱m1s;.0\*q<NvMwٽ ?'AC8[ċy\);93TƟ&uTԃ/ K_t!-)]6k #tl|bi϶Њbr@$jkwFZyr$y$Rm*I>^v&݌.jH*%5 I%]Lf;I~i HZRIn~iܜ 3r D&^<>rnڈj_RVW7~{FMy=c:T l(B+xڴ6q+҂p=&ס`g a/șתd2RZnu oeUjVoJ1LOJǀLh'Xݽ `1[xrfq6y$i]JWx !riX5X:=b&d D2U)/a5~y qlHb/vlN8it%>1F\uH@r/0r &Aa |; 2Alu~X/H]ܨb蕡PLMF;f^S)>췾뛰jDzTR#0ςY'Ǡd AB VseggXhOq)rL ?yEd;7 ~]MFECSc6Ng-c!.JD|iX2\K@'=FJQ)ݧpEc0B9G}Uj?F_Ý摩PJ L/Ix^2.yx^W/,t0la:_DA8emxyqD5,08H{V FҜҒmEu5%𤟰7-sɄ Nɥ Y:(_ΫP'ZVZ閭U9ydZf-툞fzy#kIځks&κ~77bTo۾oߩ),+܅4nr^H~:1 @[\/|0#}:gOJBvɴ;HMV^WНpZMpW~ ɴFJ1l˲V;^,? H)&!8*`/ÍeXu)lZVX,:f̠WS3_uvNNK|*SiuW 3Q:,AR.yku],q!|qDLYk>+Pi4k8Zna{@+gdrQ/&лͩ/W4(9HɅLXz ߖ.BZj3q49~/܍B]~)(2 \WAE;: aRe!VBf̠_2 셔hjIf)4ޱ~J7`l OURޜT~; &y p6xFK~1H'!W}fd/ĮjKSIZŀ>%ӕ)f y=Os T5QY> !1//`>nP;%~ID]İKQmgSmzt-ki7()0-iJ͵Tt9_BO.C80LH?$87pPa+UDڈfJ$ihu*6U8eų+Hirřb+qLTЮf8u=cm'$j#?<sfXaIKJa=b4HS,myÆO_)~ WvOmn,C@X;E!YN=xDjf:w30X`~њJXA|UsP>%p=!ʌ!ZؠL$ >FJ]=&K 9_/pU{?C"b)+[ R8{6F:~' ' o.Y/ԨKw߅4VE*OJgS TkB8Ɇj8b~ÚhɂY#(, E~+k,44Q3*f۩|>ؼ? 7?1gO ^+de ?;z T衡\"&D ZL 3 $Vt,X˥^6I*xC볆l)Ufro˒̗͡lf}*sn -_1ͭ!ou =a!O"o\8HZNr{ $N*RLr Emпh"?9Ԡtta(>ù-׬jJciY&ɽ0:jp e[5XҞw``mw*.IzDu*rŅVch3@y΀0~d~ޝ #5k 09?]΍׀na!-3S_. eM@nYsz֜O݌YOQ;FxqcLML,!K﫞ZHƿAu@FP SM,SRafKj>'{ !j[%9=U%AhjsڍqؗfhkyAC (h`Y՚J`*R`djG̠up(9Tl}AP(TP{o,/#RVHQSru@ݩ~ @`ZV= yueb~Lk'T /g% })`J[TTuBe%~?ǁCgADVYuԣ lC SzJb #XyUT6kuZ%LѴ{TkdV} n2`'3.s#yХEcgX1 xTf[&}%N&7sJ1QpeUqAA eUT vaU<#xU.>G}*y}.l6OBr{ɶ42Yf{sgxv;pQ^JX(}vm@V$<PXA`=ZtZfۇ{/wwR=[!Nk\.-l2wϝ˯3"i iAKO B;pva.0*[K8 4k 5 #rU bڄ`R$wVbNNz%!]xЊ\@!L36OF{[|?Idی./۔^rQABY7qmk[:4}nɞc7)`h7l>OA((6_CDn&^`vȪI<CDFFb#TWJˊagHÏ2VF:nsa evS|pԔMfH?O.T6ä*JtˤW3 >`h[ڻiJhubE |ɿPvq1Y:e߸䁛Yp򌛍\/(kT9y`I7qh J э73Šamo D`rS F t6-m .I>Hoc Ue-#3YiΈyO+[Z/w @ R;E5pNGbEяHN>3'AE9~R4}#yNz)IP FPCNPt`F@5z]'ge[asyC{. E|<"gnϠ-u4+e: u'D MF^˿;CٗܢKVƒ 'aXm=kkѩdfXt#8 e/ꑱ~!s?Pp,< Q7 O iߎ:A0O)a<;ڔQ.c~S5@ҺW`a /Xzw Q҃ F 菟L5gG$-ör ma{si<UJ : ~`Bt :Jw60G%6jη-/#,VN~[RECi-3xKG7^YJ+_H/*E%raF^X5(l7!s:I\}5=&yq$sރTn{,SVhw/6mF%X%J2 (vACiUxih.E+!9>>TJLr 6ۻbԐSq뢞+a:i DzV@ڙmlB P[뎾#7$O]':KhL ӋyvX({N BY;};^o{Ϯr筽~ {/>{m] 谿n[f_ M'+7su<^@>Vd7r i~A9FH6cԠ"56C SM :6t/}v&G70.vëz7vaR6c}׉I%^]qs86  K,7@zsIpf:Gkև4e_yjT.jA$S@DPS2fLG}#⬷bVx) !"lېzJv(f~=vN h>永)6sPTnTn8pM[-OhwAK&k:JL!RڕbUT3(D ΘTt0PMYhk#{PAXfzm רXd[O"5ri mݦԹiպ hCkw!MKzs]*to&?ġ;WE$IN0MP wW2ke7ClE04o3鳁^`%gArI8k9"8_ ^r|8+˧ 5*YfR6c5W< lXI7TQ_N"(.bDVT&# DIq|TΘvfA`q:=ɡڤ+!6b ETq.AxV4U=s~`Fet? mf]Aj$OfcBC6$]9e)= RGYDNl-&2>=4CDwtZVֳ1ծz(Ci~L/j)Sxُ{6i} \7LD554ʥv+A J^9I5x hG0*7ik_Gˢ`l7(4Śn85ҼE#VF*Yl_!hRm+kqz9(zs"=CR)z.:^Xq?w? į7H*~$Ys`g67C yF@$<`dyY5M;&^ˤ&4(fl\vӞM8h>" wxpXBD͟ܜ㰔BG|'6?03`9IC@DoJI>kJ]C`z:}U`W8Ii|s' ?NerTvW}:@I#ojKgFA \Xt#.(s;C@ݔ*{])7y  YFtjH̖ߞSO !ƙ$8+`JkVtwN2{V8~*g3>۴4-~(g[zX.9rpr\&k3˝?(zL򯾌x?ߐ`1d#S Ht;Erfq2{z_Ʊ-]Muњz`'Ref߷=m͙5GM>/4>S$}r]v0xte^am!+O;&wŠ5OYn<_t!(cQ;OksmB)3* .rq2i%rA@CR"^-?Y9V,ytngjrFEB[h lf; }F<I#?61s̻ Mܝ],r݌o1|pGN[xcnۮCCS Ÿ F:% l0{z`U'l {yv\q a2sAIaJ?1|*ToC#W i²- J%ȯbݟUI;& :f`EC 7qӏwpq2f^`VXsvuyvm1lG)ԀU0;]ɤG.sܿ7t$xt:l]h?s0 Rw))O@K+_Qa.٣@0N={&0Jն cоȬ/ӳhWTQҴ 8Kq'FF~+X vjb `;9CJ5/ع.$-!n+xe9%\wyj֗~։s*>? +3 VL2NS j(Gu>&"gvg#H'(7@RZo)q9H idHQT%|P[;>QC^(\B)"۹@oW@t&;*+x(-MBߡm!8R {Z^)T 7?&-܃.^Z*v Dq?0 HOd?⠠AR/Pُ { 8/My@ŕF_@&X3͇AÉP@j>pmz1)^ m wimjH"!Ra{RMߐn&G<\g+/ O_~:,9o:u@>0 _ŠG쾾H*38rx YRU'<4!B5srF6DOK Nw(cש7 }mD_#ն\ɪˆ!1WGf&hf9M<4_.iե;iٶ,1 |-!iO5ՇP:y M~0,Ha--M(8Xgžǘ+1yax<1/ q4oc.⸃ޜr+2BqA.+&K @, >uP ?Y-1ŷ`U '=$`+ڻ?ca8,EXB z~-w-DʓX42{AZT7.f^rbTV$Nc%x n^T~Ou( QrJK2@ܴ)CRȥT=Re!*aCǛj!kYGxCAr:$QwYiN8{Z"c( h]{bPKUk^iit3nT `0zJ|O+yu$(`Ʀ Z2ouF]:ȯ )ЧǝdjMJv'-ƥwHG{]{KJZ 0yrHw^E̡ RQP !Q%ltBS σ-v1 jٱOh} :@2_)lx)~5xǏ`AY DBxJ h\61G݆*q)3!zp4DF('7*ྺWAA%0aJj3sϺ5:y-F3`3-u?_݀FM ZS> RlMG|*I7buĠW͉mK"oiG@x+qa1/ p͘xx 56ƀC)p8pa!2M'BkUĺ7vÀo@Z2 Om?c%6pLt(JBXi{at+G{߰[&2Bò S3k`~ui9u\&Dա_D'q_Mew6%yR87¼}BN^Kby*?8y<54r%vB h1mwYSDGsl@!%,@!dH"/ʾ= `+XMQڒUwrjضYoKMDg!@ [=|C]l(۪rF3΢s-d &.X7 MvT"/ީIJ1K".py8}s).2w:ShN߿i5[Q=asaoI 1|ikB_OͶ $0"U;{}\ ]@ Ne.E^d6>2 -{]k dDX/#bhjs֘hiЮ XeT$sy=eNJ ?m(/PR35:GhY[^A(V;ڀG!-.0 a#r B'r J=B?#:jn2zݦ8*uxEźa(-9&X9| dPNEk /Pkz+} YtSC¡w3@100~TTl0Af`fwMRoM' ?G)Z}:mùE;l%CF -lUb-+wnv=G_Gd 0>KdjEJ:cSx%0=˳vPǕ0= e! WP!#?׮>H* ܧdzj_#؀L"=|\VR%E$*Bxq( z':վʶϗۈ%Wh'L/@Zfp0@ ~>wb ƌvZ)Qg*0=A d}Yl=p[Wfٿ0P"u\PlzF!*i]|f?s"#<F H KjF!w9#"XH=U륜[81=2Ƅoh;{{QSElv՘z03R_A=vI*}H m kp'O8J4+öiHV\/E~h]Ue}4^Eo~|LNÍM!ê'5&uSbZؖI'v׌^O/9GY:Q@F3ְT,Z!.4a=aePG yt^#Ocv >bpTͩ-֪ufe>GVz MQ#h5-FWww2jnI zb?r_?6 \LAB|´W0^%0O6Q>:i"^S{L+N\MjlW]սeGO &;TiȐ =/0u y1[]n.#˜Ƽb7q$h6?IWDMf|o; ;*zjOzMKm>46K $6Y!Iw0N ڄ["2}brPUC0`nhLdӕ%:`nLq_5~II!tXOM? ryD(GiB%*,7'4ŠwJ<ܯ6(nC+)c:ig6{W*̕*TPe"i[J#sq-Naz eydN0 +ޝK: *.;etKI+``E0%uKAnEO^ QV)/s:GпI] p8v=2iI91|MUF)ᥓSӻ=eToE\g|aLȕDLlcGEeK|%MA_hzdLV 9W~M0՞$D"C[z JҍX>:t~FXWjc>6)I1UU5fѩ`"2C]a:l0JYhWo5$- ۑBF{׻RnsIԳDkcc ˪WjKʡ1f2mPNbw)OO%ԯ,Y/Ŧx `"qh >nM?よCD<+["K!rݲ;7x!$sa 5p(>B *x"J A/554ng[]L[o9r6zkCs>v̤Q]Yk iDrL7bIsսVVA27!I`CD2NtTiФu7yZ{ 66z Dk./&ٓUAPZd (*}FRefODy*sxIgazp2'4SFF;CYWmt[ 51ɠB?}<7%&Up/=t7v>_0S]$r|'[m{ E1TXr,J~8_(Uci$2RL|;n,9d/fhXjTMLm49u^?y ysPZ Gkι 6V*y@->ܻYi{-Q/тH/hj!%#Y_zuQQ"zU2$=ZWU2M|Z~]aٍY. .X C+ ES\;?~DcEl_orM )b-kjff$g#9Ef,7V]!gDn1yQivߜdW5vf1IZ "Ot۫>ʜ76 TkGu(k`-Z_z5͜:(&FNPJK1)daIyPmM'=C^55R IEFUKӊ$zDFe΁ʛ(H =/KpN73m>{.uN)R+4b)IFymU=Qo#2sM$?=) !OǧHQ<0 Q`wZ:@HGc~l|C NW ;wE|Ԫ_W;) q\_`YL߸ENP\imb4hԱ6A-cR#pY~r%!zkXLB7ynSkJ"AET.f+vkJ%@ z @}#p0c%"pLST@N?4^Rj_cB()Qqt🡺P`[B\u"麯=9ZTuMP }h Qj덚#*iMڼBn{;(vCM-a2e VήF Q-%L7wI 9WjP%Pb7@zwd(諩rjQI`=lgk'qϵoli|fIe }Y>O`Iè]./jw{UjgrD;\r{)RM@V pbcb 8ɬ`!s52fי ÷Oq'm;_)$(PK$V2-5IrLsB|g}BfǎgZ@:WnsWir*@r]œz 刡_?2*H)qQ O^~*gxx}T>\NJQ%k9p@Xq8 'h 5EN$2"}t0  hАWv}.y\jhܤѩJJЉ &3]v^Vs0(8DU>FyI\mb4*~uxўL )^uK81*> 㑊 `m.zɤ1_>yCGox{ K=tnY -%n+%n1٣W&dՍyM{ h$^iG4eM~\FX oˆba\0svy}[FQxDz7A4 &뗥C2@]Ak/5Ex χ_*Ǩ!umY9^X҄Z ˒]QLm9|mӬ3z7OxGL, ãT)ػ\}FǑvBl}9r#T#&d24.qʎ"1u {xxo`~Cy'r=l bٜh4S_p[BDD-3[Ҧm?f~;sNl^v>yqHî(T%Ԭ^[)19x)U8( Ov3GW.Jy7D [9;ms㴚5\MNh\K".oO`Lriׄbd|mm8 S|X!e=us)[Pf\8}-<^bz]d&z(̆ëFfFU>8P3M~YVW+(ͤmhkZjnIW+\ ļ^.q>G=pv}0D6b$H*pԖ?q3r 9SvYz:6mf8S`CJ9ܖ$PaQZanTCŹp;O2vU`eYEѳ-=uTe|~mxt1^V>2 0j:@i=-h[[ܪ<ضNS 4uh<0ކceu;GYvLM#7ʻU˜OpE;swU$縋|tYQCDYGhg$4R? $Cǯͳ+I薌hO-"4$(xqGr#gOX~?|'v??&$9lЎp3zTLCũ/,dYW]bǹC7 2AD_8i"A+V hr%Hwܹl&l jI]sp"XџX/ )U =v $y+rv pJPS v>8'TZCOP5T+AWtL%ҰQ54912zx.ddbMeC& 䋵p0u$JZE]!U-7g7Xb {&Z"H~çq^X''f8z17Y!SHL(r|b4wA9ylF2xK&0X +t60A>}ob&ByMd^$I5[g+yh~r?/|9L1Qu~t)TߑgN-Bk(z ̯ۛa׊iېQE`i ŎƑQλ;zmhᚔY*\3YV̬7*s;ʩKHPq6Y՚ P:ЎRilp1`L@-mJB-:~V4:>cU.R=Rr𔎶K=CmƵ Ue nGRE~g_@RElZ),FF9ZG$56!RѶM"jN{y4K5uR&f0?&3BVqB„Uǀu1zC!H/_ Қ?D*wIx`|?~{I7D ULgmeR#nug)Dgiaojx33mwkh#b;Sɼ-&zߣ:JZc;>1]FHPT2hk$@B>lVy(֡߃"R6qqD+0 @ညX#۝,8)(mygg"N]3g楴Cچ``bX*E3 v?HU |R>Ⱥ^s2!*d0T8uب>IF  ʘy7MeڷHXޕZ=[zUC< \:#:F2߅U9b]8NPa((X3٧߅[Aq&WaL4iD\脶J(V]ct  FeAb}p䔣:+"p̛Cr.MwT[ w,ZHV%8Gܾ%Zm-Ym(+@EC~oop)9Ep^憧;,(⎄$! OJ4Kk31 (}emxwfwGrIj;ZxXA G9x;'LO CȍޗooC6/?gN ^wZ}ф}ku_ J$./RHe#:B7~\eRC<MHR|Dj}Z%d+%Ϧoʘ\5z e:@!2zb\q 5n*Z{n0YђKU0";(㥮뺫R_ 5kϊ*#2 M {%O]QtJ P%y A]0AiLƿ/~ uކ r:uX74(Iy>rE`$EkyeP'nI_o* }kzUQ Λy)DJdebj*/uۨZ H!:QL+:o}ɯeL:a{7: #D| D۹)E yx5.Kwact=Kb,8Ny@Qn>rP,CjȊAqZm4);DA2<"4? "/\=TB2@#YBI YB=BLЅs?TA^6֊Jo23D u!,_rh'~*u{x' NC4k씾9yS(}hLA*o 'K m<7߂3qF_E^G_*SQD?{w%>yB<Ȥ0>5u' j-&m"8|Po(/> F| Kc:z*]t9yy_f2 D*~NvhősI_JQW MtL5 }Q5NNez> \Fd 5J%*Fwڲ10@|(O۟WӦC,%ą^wL+p%jkaB ލ?EAyǑEL(w9Tgm[ ]5z=}2J aZ(XGkhGl?%(^u󁖣1}I?>RN90/]Nd%Zl*sӸÓWk5;3MeW|Ve8WMT%CS½q eg$x!ࢊ'kyAD$u`gh{ ;)ZRަUc}G|sLzI KZG\9\Oxc}qkǓG%-Tfނ+Dv^>鏳SeHzS_\{E9V >#Mr>fm66 KfaG܎7A؍~#?5.p⬯'J5Pd**a:|f>K-]= O\&R<=?z'R.Crw& ?qthQanmMJ; 1'ZP&so~;+4k IIm:OX7ٜsX]l_ar =ft;1@0x歛x2p?OV\..͗Xb]Jr.QzɒkTzt\F gZt5޴FPܛ갺ABx2haWɶƲ/< 4YJxx}Po}H @u^,!x[49M ~ڱ7|0/vVi,= N/dKm s*T# !ޑY|ܑL^T\%d{p0sfl/QWڏᤧLT|yaKZIͻ\Z;jeJ?c.S kɏMH ?ue9bQ4Ū~+!sjTچvqՋר~mϥ Bo~7qwIm\!Irʵ|EL^&BB,"FٛYmݎmҺ:}h<1V 닖 B $}":{G9ą%Kj$U[1ô)'$:P0 ;ImMQQ `eX'h8bSL>mh/\!^3y\3uٔ,Y}|(n,6'}Fp[}JUke%1t piԳ\@=]>#==XϪ-Tqd6UӍF~<=l0xe@J$(p UAe(M1ibMϬT(!CPXQCMަt;ђl9z Š+aeI'Q.DG{bDm|õ]!jaA&\JZRV?fr#f/Mp^,u7`$SLD38°]댽isŬ slISo T\6]8r®K˨ρ:z : <NbbLD;:-|.j&E'&مHG.,.S8;FԬ:uN;DmT鐛TkPrQjERC9.BU+pf'I 4tl,c-BaUUnL7`3ք&VBo0Ib?=MC5%7vf~]Z{{,Ij=Z`l ĎbkxRԆm/䕪p!< h3L $\'bG[{V,HLBݻu!Na*3Y, @Z3}uV AKB^ KM5 o`g)N7iȡI };n.7ӌJT,Rq(Y0Y,*0篧8x /|@3p*-b% a8iݣjɻvh7- f>(zEgfIaQ`X%{+<)zL}pq;NSe;s ÃQfW?m0G4}B߽!`!,L{0Fhd]?೶B)~ПcMr4%?%"%Z9cmO CcHj$u,˒d Xe=PI]d6󶥢r^A8`xov"u6a~-> F6N6.թ|1y:_X9F,E2cG|wZ"鮴´n$y 2T^nSaXtg=5IP)6`g7IjcJDW SZ޾>;W,.b68CD"v> c\іd6JI+{wbNEζț] " hv쯿.E=LPōF!-.EBMLhwiTh&? !Ғq:˞ڻtifpctCaھFf %0m=2L@3wM94g-W{kQn:(^8bМ) Fw.&0"0Wd[1wNS;Aڑ,zZͺ`G] 5u+وѨoULyL&AH?CߎjL'$\q> ` Y( >-}m!h5U-$UhGK sBnn 045 ף1^$t]f?  ;%{1N11}ݣxX,ON)V'Ȩi(tl,gv?敂tʞp'd vXL_RkO^k-h/`Kxg۶l0`@ujM*8{`!~9lBLOǎ%jD8rD9uC>hZ$iE12Ć2yʕOP0gXp\;Q3.w#rv/G/k*{okÈ$q9"3bW /&3(`qET+(z=%+xnUU#!68KxyU)`^[uΤ"'q4[]&>/mR#e5=v*B[C<)5*7E^߁Y\YMɃa'=_OrхWƣkBb0Dn3-pLTC9CLLsTavAƀ'U:4ɫw3ꢖ4A2lcLkr$c %$l)%GR-SyۓTZ _{_4 i^#" =OZx1M4PEΓz*6 2?<BkHFZU sRp"ЇOȋE$~YPԒ)4Z0" AWE(X&:C#'v64WfbWE\2EVQMAᅮCUlb8**)Z8wxgQ(!Ɏ݄|st@3=W )l P~Qڷ,6tWc0 |uv34/aeCUD zn3cAVաvasflEb&3X nX @ w>.T`ו[n<'U/7>u%b@-K C+Bzxx}7!oy;Yd*79u0Xw$g8P&fO.w夿[!YFw i8wu1T s dK3-sr[8oD=wU֮6D*xyE(hzUN9#ݤ/PD0W X[/RbΠ9R6eNirj݌,Z@*jF/];x_! $uRi~g'I/45M<"~Xm(d^9,p95I)-adݵJ:MIMU\"VESЁ̮T$ƃy^ +3hQ۹A"":lS.S PvKYB3|FM+a`A-(~/"7$uBdfo8MtJyxsZ ^EB=(SHSon+ fP8tAMz+傥-BAdPlo݆}7G9wXWsoQ>30TYEGl,[D5i#H t ,hALN.=ۮܗG|l,ux#28{źQ'^9TE*߳hL481~C)ZgG䔐ׂMh#%掖b xW DD{|Ky|#i$-W'* & o٨‚-cw23AKݦp?N/TM(&+Z@B fV=iov\Ae6l'*9ٿ i^'!,IS~a鑫]R+{V_neYF$~-˪2$(r|A 9a h{!"TWx'u\.q^'bqjtT *BMجD}2]F*Ytc%]``jX*E4+"AABG#,n>K\h?WOWĀK%P ۽{ԩ |/0 pUشqB0s0=B{A+Ѹz5v*OYxX\^KJmwR+//G*Y沦Q BɺhK? =8۰)lHfC1Ӳ ^HH;w_T0["θ9iG'ql)J:W.Y4(Y n)SԊ) >H:1dq?l$]Ry98M"$C`LhY9@oj3[Q!q*m ҬeƉ @&&<"Dqf^iddfLR޾všϪ:t왰̆׍#~A'Z+MʀAD6mZ(g'Aٜ cgGQ2|N 3A*è g;=O#4B:%0:CIԆL"9FBo Xn#'_-oJ!x7;i߱Ye;?\v]~@o=:TS 7aߜ+IZ\1NaV t)ϳHQeVѱjq k )!X WF(eC?=/ uJs{ r*e ݴ=0~ߓYKOԱ?zo%_?fC\Lq bv?.$t~梄uЏL4GS״Q+x ?\ٝj#&Ա&s;Y*ҧ=C \(?/JǾK5DF OJs{|ā)-:0(r-7矔J^6QjKBc][t# V c 8+^;Xޙ `k1|%L+bEH?s>XS)$|& Eyh349qG/+n6%\;;G 9D\Ϲ^4 &~\'Pum1gZJnwϦ)K~CX8TT`mqQ|_Ey1͆6Syn.,»D &@&}sEB8bXJO%6/C&D^,( \w6e- rsC?~'j^ܻvOz; Zcvi+&>|~1m/JX/m|O b۴\_"TD&n"Jd-h35~-X9]s zE pCX1 p7'tvFoՋ9x)yDQ08`YHǂV#>^L2Y'`+i1n ^]tDD),r΄=ʉDZatEVcǒE,Kb"YzqvI.؎k?_qdXi駑G˔*wAR.l.|w+->^kA,(g?:"[A=vMfﺄ4rԑ8/j0Z|up/#T^1Ud ];Z]~P>FY(w_ջ#Ptu fk_$LV Vv!>] LJ}\AI~CK|bIձVG= ~=Oud`v p(UWU3{~v(^+D g;Ɗ"mFusC `T`t29obcNoL8Zhm)Wzd__H$5SLZoSn@&{IsعI&oNtyT]Qx/BG:^"ȺogYWz& 8jKaR8o: Z҂36)eCp+s%8 Go#vAacJ?ݛޤ[/5/ʉI$%'qbfMnRۣ [uێN4&\,nBїes))r"#E yOVydM'+MQzQVgbu;'އn ~j:C1wnDJΎGӺEA=W7!-*c[K\fa%/+r/Cpy(b/a8T˟:" -Y9X-tfXAm~S[-^_1 - _7;K_B7<-]=6 YFjb#~P^@r6vG-+ %z 5D& h+G2@lę2WzX?u6q+SuO_@U-<똍[rx-UiЌ&F33ᳮ`; P^'` Щ Fy=*I~#rj<1p A5 +@ԌѸA`^| yUgk' 0}4<Rtaj8h2/9lo;[/|wE͢qFlmעw4OO1ñn[p"āh1h/rꕘRjixiVf`dy۵OTd& xq\ KmQTn]qٔT# '^ħW떸 ÎoUGٍaI*>M޲fк BWIdJ'V_pLr /+R.r!6N˟J0U 9SÍ ?RrVi&GuCdm^:Œ u85ݘ筳C V+0'YY =slR#62h|3ܕ0LӁzAIw/y6'~nHa˵3;Qmݷ-Vrka]k=nIa$<}Ysj׺FzZ},cW7hraNW :,<`썵 371wfLA"w6R_ݻYÌFkRhCCR>&Çx9QiLnlzW# qGcc^yPD\ȩ?l,5y {'J" @ǧzf: Y9*}79s/;k^IVb=Zʹ27cxh~ V ST"̿5ߑKɭVt2zaCt6olB S_)_IQP4Z(ϧ;xýjќW-o4j! <{$ [TYn v7Iz^C"nnҲ.yHtQsX2D$9\W~Pz7+x2mpm>EmQ0;Bs+z (.OI/*>L.}ҥx|c(z4n/lWP_KU=Ծ`,7•IAjִj+ gj?i:s & ]5sYn?G;^o攪eUO~7b &Y0096bvbĒ=HХ(_usp$'cb˙cScjn' 7BR{X`+$SWUYb>Ж915KbӟQ|' T1i}UO9 Au}W6A4x핅t$Ȃ2ʎbBs?܊:xbmO 2I8yL b~tM-PO$]5 noNW2[Sl\! a~yyV/ mu{<^^HX.>.!wH**lx&N/Wf˼ց?N~|&.fkS0S!u>绌P|%=b33ޖMF([>BO O[g3UZcv}Tv u٦|G(&a`Uў)e=ww@)Ay8ώ y`ª~ Jq^Jg%hy%HqFTzw<*S~-W6Ws+Ӿf+ۯOuHw4v^ec˟P@Ȃ)f(]v꣉>I%3`Sx햃ltF1H!(  jzUeAC{ *nyll>)|\ JoIYPh hѺǐ5u~FOJe"kA4dF.&ݏx↌q吞p; "ۼ YKyNMغ> ̲%Ct1O`VYّZ"v= eϏ?ZMbȎwCƻ"I d!KWL vøw6Y)()w 3K / =79ac3d"yPho(Q={p4؂<|'ki *ROI>A\2bd#OkRҶyg Xkk,q{tdo5- 8(OyZnbyrRRItqĿ7,>AFlȌ4Y_N>^FU[lz7Dm4-?~l$An6X#3R]m>8]ƁѢI]ЫC ƥ*sWl`3XǭMRQ9s]%6yǥ[ialBnnBU}eFU UDS^,~*I/5fe,DSW,dԚzCd]%̨m7UX',uX ybMLCWot,9*AOF,:c]a9LgW)+w}vBΉV]m]ѶpwQ|kߔT?ɉhph^e.Y 0 G&Q} _[l 3oSANU#swN%d4ծP퓹!):c=:j$&vbѓx`"oR[x NUckV.SpA? jY z?elv6NQO&J [?fQA]: "E|_QTN@xYᗚ2փAvjPckWD@U.t-HW:j\XXA ߡDݧoˠJ_I) ӺOw?K$Id!ӿbI'4ޣH#rZkK{s7-=/s7)j{PHPV`_ Z~:E8fMa]EZw2pSU^^dtؓPSՎoW s`g&IVrZZ)V="MYPV8BCwwK&H]F?-?I++?4 3y?{S:| ڷ߮9>G1:|RB*~ܿ9ė\,Fjo,RAs(gYªDrEw66161R ~eX#~`Kw DnT`2. _&64 c=a`znKJIܝZD`[528UMVrȘ׫,ė)A{KfSZ(ح6.!5~KF2zz.,c ޯ $6=) g˵/.,?xu}MTBrS‶:@؆}2e6M临T@1g9B]v;Ņr7>iUVj;HyxE0"R#7yσTKE<]i71\3豯8|ImO4v^)-3S󵝗"5 &4~W:0E*YKMW![YDxmV-64܋t Z${4CFtGJiS⑼e^2SSH#7|g8/(#N3?+{IU h{—w&Nd۟F8HH*2h O` *+{%QrIuҿuq/N(I/Cl#7]EŽRyQ^e0nأ]M)#c βN cpl>7LJrȕR@kq+}횻Q-h6ز1x)ᛖJe/Pb 季E*hKa/cq)4 #-/XN+PCm|*p;FiHL/;zbZDFnr̻8U9@s._K+5F.G l2V\J@~ʽ;_ca_2W$ISA35myE")&cXty[QAqDjC Z)lN4lX1ϲx fm7iA)h(N@^rc AjfJ4^*nQ8Wn9Oϼ>05l_wa8@8XR X )py}/&gq30M"AhrT=>n0%f)iAeIRЦŎL,xLJFK`6"+mЎDBnzI&[fɖSBSTg@uQɥ1*a@}hnÇC?{.+0&Uz![g4Bgj+I,PZF9Drt,u~ #@09xaJ_Oog-R 0U9=禥;xD[:IbJ$i(g޴W CЊ$=X1)e?O٪T>]Ur>orKܶA?*)U:H*12^adߺ m+SF۶aw+n).iCZzz`KrzQY;a G+Q н_"HΰAQtut:؏; f zMu׀AX(uv3CӒ' &ϏH#r ߟ4 !]A)W˂ Nvq':KW*O 箕η`n< 3m^]`D)p;xsҦ 66H!,! F}=MEwP#PA"*C|QbD[Tƹƹ{/M7Vc4qS% bC]#[/St?x+FBaQuÆEK@ +y]ɥDەtWbŞOYۼqW Q+я O)"ǵ6LwNW~yj nϋ\k|hj)pZ˸ANk3EWl@%aKF)Pd$͑ Aܟ*MN.qټ->3ьo t\!qى^*K}84}&꺼U>ozf 9cf{7?g#R敄!D(kMÂ_/M䬮8%/gZk\nԣVIu1NȒ0caRz@>3AOZ$Hb .N`TfsAs@HřFgίDe I~180Dsˁ&!bUI0 m r[Mm//߄1L@sً9M, y퉗B K#.RDma^+qޏ~0._gņ7KMGXQp%LseLԧf ?R>OW=˄2exU[oε*8s)g c%H"ZҒHõNhg94Frbg8cgũv>xCNnf*ۨѩ9N#7,J!"r"A&ow\]x=}bWEr'U⽂$tCKjMn=|ب_xd9ypm鮖8'0(&oXFu5:EJE!Ɓi56/!T0!Zk ]ⶮ&Ydj+z.Efa;9YاH9ݏ]FfKʥfǟP-ȐtN0qX߅!1k>s/h:B6xE,HDnCɟ!t)7)7zGTτT%4Q88 O3o=+*❕uy^TJ, !jy OA? oһ*͗ªՕ6!IQy%fA\ zBmk) Tf]@=h[nSTN}H(HnQˠ{)_)JV?|^c1Qa pX#s20=ֺ\%(ҴV-%h~z{a:(3x@J&!,Y֪Upsjd|:U&XH`%v iD҇!:{| 8IXD3UD"C0ג9߭ܚ 9fΓLˈ=.7l-ϜIѠu]ƶ.9[p YZ