[asterisk-dev] [Code Review] 4112: testsuite: Make tests/fax/pjsip/* depend on chan_pjsip
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4112/ --- Review request for Asterisk Developers. Repository: testsuite Description --- PJSIP fax tests are missing dependency on chan_pjsip and res_pjsip_t38, causing them all to fail if Asterisk was compiled without pjproject. I have not looked into if these tests actually require res_pjsip_t38 (I don't have pjproject on my system). I added it since all tests have 't38' in the name. Diffs - /asterisk/trunk/tests/fax/pjsip/t38/test-config.yaml 5649 /asterisk/trunk/tests/fax/pjsip/gateway_t38_g711/test-config.yaml 5649 /asterisk/trunk/tests/fax/pjsip/gateway_native_t38/test-config.yaml 5649 /asterisk/trunk/tests/fax/pjsip/directmedia_reinvite_t38/test-config.yaml 5649 Diff: https://reviewboard.asterisk.org/r/4112/diff/ Testing --- Verified these tests no longer attempt to run when Asterisk was compiled without pjproject. Thanks, Corey Farrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [Code Review] 4113: func_cdr: CDR_PROP leaks payload
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4113/ --- Review request for Asterisk Developers. Bugs: ASTERISK-24455 https://issues.asterisk.org/jira/browse/ASTERISK-24455 Repository: Asterisk Description --- cdr_prop_write allocates payload twice, causing a leak for every call. I've removed the allocation with the declaration of payload, leaving it to be allocated at line 563. Diffs - /branches/13/funcs/func_cdr.c 426139 Diff: https://reviewboard.asterisk.org/r/4113/diff/ Testing --- Visual inspection only - my system is busy running other tests and this fix is very clear. Thanks, Corey Farrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4113: func_cdr: CDR_PROP leaks payload
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4113/#review13593 --- Ship it! Ship It! - wdoekes On Oct. 27, 2014, 8:03 a.m., Corey Farrell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4113/ --- (Updated Oct. 27, 2014, 8:03 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24455 https://issues.asterisk.org/jira/browse/ASTERISK-24455 Repository: Asterisk Description --- cdr_prop_write allocates payload twice, causing a leak for every call. I've removed the allocation with the declaration of payload, leaving it to be allocated at line 563. Diffs - /branches/13/funcs/func_cdr.c 426139 Diff: https://reviewboard.asterisk.org/r/4113/diff/ Testing --- Visual inspection only - my system is busy running other tests and this fix is very clear. Thanks, Corey Farrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4111: app_queue: ao2_iterator not destroyed, causing leak
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4111/#review13594 --- Ship it! Ship It! - wdoekes On Oct. 27, 2014, 5:53 a.m., Corey Farrell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4111/ --- (Updated Oct. 27, 2014, 5:53 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24454 https://issues.asterisk.org/jira/browse/ASTERISK-24454 Repository: Asterisk Description --- update_realtime_members doesn't release the iterator for q-members, causing a reference leak. Diffs - /branches/11/apps/app_queue.c 426232 Diff: https://reviewboard.asterisk.org/r/4111/diff/ Testing --- Testsuite against 13, compile test/visual inspection for 11. Thanks, Corey Farrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [Code Review] 4114: Prevent stringfields from accumulating unused memory
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4114/ --- Review request for Asterisk Developers. Bugs: ASTERISK-24307 https://issues.asterisk.org/jira/browse/ASTERISK-24307 Repository: Asterisk Description --- Any time a stringfield is blanked it currently prevents any currently allocated memory from being freed. If a stringfield is repeatedly set to blank then set to a non-blank value, it causes new pools to be continuously allocated and never freed. I'm unsure if the loop can be optimized, maybe the break can be re-added to the original location on the condition that ptr == __ast_string_field_empty? Diffs - /branches/11/main/utils.c 426232 Diff: https://reviewboard.asterisk.org/r/4114/diff/ Testing --- Manual test using https://github.com/elessard1/asterisk-lab/blob/master/examples/lab_stringfields_leak.c to verify that old pools are now freed. Full testsuite against Asterisk 13. Thanks, Corey Farrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4112: testsuite: Make tests/fax/pjsip/* depend on chan_pjsip
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4112/#review13595 --- Ship it! Sounds good. Note that the following three tests also require chan_sip. /asterisk/trunk/tests/fax/pjsip/gateway_native_t38/test-config.yaml https://reviewboard.asterisk.org/r/4112/#comment24109 Also requires chan_sip. /asterisk/trunk/tests/fax/pjsip/gateway_t38_g711/test-config.yaml https://reviewboard.asterisk.org/r/4112/#comment24107 Also requires chan_sip. /asterisk/trunk/tests/fax/pjsip/t38/test-config.yaml https://reviewboard.asterisk.org/r/4112/#comment24108 Also requires chan_sip. - wdoekes On Oct. 27, 2014, 7:41 a.m., Corey Farrell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4112/ --- (Updated Oct. 27, 2014, 7:41 a.m.) Review request for Asterisk Developers. Repository: testsuite Description --- PJSIP fax tests are missing dependency on chan_pjsip and res_pjsip_t38, causing them all to fail if Asterisk was compiled without pjproject. I have not looked into if these tests actually require res_pjsip_t38 (I don't have pjproject on my system). I added it since all tests have 't38' in the name. Diffs - /asterisk/trunk/tests/fax/pjsip/t38/test-config.yaml 5649 /asterisk/trunk/tests/fax/pjsip/gateway_t38_g711/test-config.yaml 5649 /asterisk/trunk/tests/fax/pjsip/gateway_native_t38/test-config.yaml 5649 /asterisk/trunk/tests/fax/pjsip/directmedia_reinvite_t38/test-config.yaml 5649 Diff: https://reviewboard.asterisk.org/r/4112/diff/ Testing --- Verified these tests no longer attempt to run when Asterisk was compiled without pjproject. Thanks, Corey Farrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4115: res_fax: fax gateway frames leak
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4115/ --- (Updated Oct. 27, 2014, 8:12 a.m.) Review request for Asterisk Developers. Changes --- Address Walters findings. Bugs: ASTERISK-24457 https://issues.asterisk.org/jira/browse/ASTERISK-24457 Repository: Asterisk Description --- fax_gateway_framehook leaks translated frames. Over 1.6mb worth of frames is leaked by tests/fax/sip/gateway_g711_t38 by one of the instances of Asterisk. Diffs (updated) - /branches/11/res/res_fax.c 426232 Diff: https://reviewboard.asterisk.org/r/4115/diff/ Testing --- Verified tests/fax/sip/gateway_g711_t38 no longer leaks on 11 and 13. Thanks, Corey Farrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Dahdi building: autoconf generated configure script
Hey there,
I've got a trivial question I hope to get a quick response to :)
I am fiddling with dahdi 2.10.0.1 on non-linux variant of UNIX. To be
more specific, I am trying to build dahdi from sources, and am a bit
confused about autoconf.
It seems that 'configure' script generated using 'tools/configure.ac' is
not quite the same as 'tools/configure' (which is supposed to be
generated using corresponding 'configure.ac', right ?).
# autoconf configure.ac configure.generated_via_autoconf
I am attaching a diff, that suggests that last configure.ac is
out-of-sync, and seems to be stuck in dahdi 2.8.
Is this correct ?
Sincerely,
-Ł
--- configure.orig 2014-10-27 03:44:37.191346234 -0800
+++ configure.generated_via_autoconf2014-10-27 04:00:08.520237312 -0800
@@ -1,7 +1,7 @@
#! /bin/sh
-# From configure.ac Revision.
+# From configure.ac.orig Revision.
# Guess values for system-dependent variables and create Makefiles.
-# Generated by GNU Autoconf 2.69 for dahdi 2.8.0.
+# Generated by GNU Autoconf 2.69 for dahdi 2.10.0.1.
#
# Report bugs to www.asterisk.org.
#
@@ -583,8 +583,8 @@
# Identity of this package.
PACKAGE_NAME='dahdi'
PACKAGE_TARNAME='dahdi'
-PACKAGE_VERSION='2.8.0'
-PACKAGE_STRING='dahdi 2.8.0'
+PACKAGE_VERSION='2.10.0.1'
+PACKAGE_STRING='dahdi 2.10.0.1'
PACKAGE_BUGREPORT='www.asterisk.org'
PACKAGE_URL=''
@@ -631,19 +631,6 @@
ASCIIDOC
USE_SELINUX
PBX_HDLC
-PBX_DAHDI23
-PBX_USB
-USB_DIR
-USB_INCLUDE
-USB_LIB
-PBX_NEWT
-NEWT_DIR
-NEWT_INCLUDE
-NEWT_LIB
-PBX_DAHDI
-DAHDI_DIR
-DAHDI_INCLUDE
-DAHDI_LIB
DAHDI_DECLARATION_AFTER_STATEMENT
DAHDI_DEVMODE
DOWNLOAD
@@ -653,7 +640,6 @@
HOSTCC
BDFARCH
BDFNAME
-GNU_MAKE
LN_S
INSTALL_DATA
INSTALL_SCRIPT
@@ -711,9 +697,6 @@
ac_user_opts='
enable_option_checking
enable_dev_mode
-with_dahdi
-with_newt
-with_usb
with_selinux
with_ppp
'
@@ -1266,7 +1249,7 @@
# Omit some internal or obsolete options to make the list less imposing.
# This message is too long to be a string in the A/UX 3.1 sh.
cat _ACEOF
-\`configure' configures dahdi 2.8.0 to adapt to many kinds of systems.
+\`configure' configures dahdi 2.10.0.1 to adapt to many kinds of systems.
Usage: $0 [OPTION]... [VAR=VALUE]...
@@ -1327,7 +1310,7 @@
if test -n $ac_init_help; then
case $ac_init_help in
- short | recursive ) echo Configuration of dahdi 2.8.0:;;
+ short | recursive ) echo Configuration of dahdi 2.10.0.1:;;
esac
cat \_ACEOF
@@ -1340,9 +1323,6 @@
Optional Packages:
--with-PACKAGE[=ARG]use PACKAGE [ARG=yes]
--without-PACKAGE do not use PACKAGE (same as --with-PACKAGE=no)
- --with-dahdi=PATH use DAHDI files in PATH
- --with-newt=PATHuse newt files in PATH
- --with-usb=PATH use usb files in PATH
--with-selinux enable (with) / disable (without) SELinux
--with-ppp=PATH Use PPP support from PATH
@@ -1422,7 +1402,7 @@
test -n $ac_init_help exit $ac_status
if $ac_init_version; then
cat \_ACEOF
-dahdi configure 2.8.0
+dahdi configure 2.10.0.1
generated by GNU Autoconf 2.69
Copyright (C) 2012 Free Software Foundation, Inc.
@@ -1793,7 +1773,7 @@
This file contains any messages produced by compilers while
running configure, to aid debugging if configure makes a mistake.
-It was created by dahdi $as_me 2.8.0, which was
+It was created by dahdi $as_me 2.10.0.1, which was
generated by GNU Autoconf 2.69. Invocation command line was
$ $0 $@
@@ -4332,34 +4312,7 @@
$as_echo no, using $LN_S 6; }
fi
-{ $as_echo $as_me:${as_lineno-$LINENO}: checking for GNU make 5
-$as_echo_n checking for GNU make... 6; }
-if ${ac_cv_GNU_MAKE+:} false; then :
- $as_echo_n (cached) 6
-else
- ac_cv_GNU_MAKE='Not Found' ;
- ac_cv_GNU_MAKE_VERSION_MAJOR=0 ;
- ac_cv_GNU_MAKE_VERSION_MINOR=0 ;
- for a in make gmake gnumake ; do
- if test -z $a ; then continue ; fi ;
- if ( sh -c $a --version 2 /dev/null | grep GNU 21 /dev/null ) ;
then
- ac_cv_GNU_MAKE=$a ;
- ac_cv_GNU_MAKE_VERSION_MAJOR=`$ac_cv_GNU_MAKE --version | grep GNU
Make | cut -f3 -d' ' | cut -f1 -d'.'`
- ac_cv_GNU_MAKE_VERSION_MINOR=`$ac_cv_GNU_MAKE --version | grep GNU
Make | cut -f2 -d'.' | cut -c1-2`
- break;
- fi
- done ;
-
-fi
-{ $as_echo $as_me:${as_lineno-$LINENO}: result: $ac_cv_GNU_MAKE 5
-$as_echo $ac_cv_GNU_MAKE 6; } ;
-if test x$ac_cv_GNU_MAKE = xNot Found ; then
- as_fn_error $? *** Please install GNU make. It is required to build
Asterisk! $LINENO 5
- exit 1
-fi
-GNU_MAKE=$ac_cv_GNU_MAKE
-
-
+AST_CHECK_GNU_MAKE
test_obj=conftest.o
if ac_fn_c_try_compile $LINENO; then :
@@ -4620,402 +4573,15 @@
fi
+AST_EXT_LIB_SETUP(DAHDI, DAHDI, dahdi)
+AST_EXT_LIB_SETUP(NEWT, newt, newt)
+AST_EXT_LIB_SETUP(USB, usb, usb)
-DAHDI_DESCRIP=DAHDI
-DAHDI_OPTION=dahdi
-
-# Check whether --with-dahdi was given.
-if test ${with_dahdi+set} = set; then :
- withval=$with_dahdi;
- case ${withval} in
- n|no)
- USE_DAHDI=no
-
Re: [asterisk-dev] [Code Review] 4115: res_fax: fax gateway frames leak
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4115/#review13597 --- Ship it! Ship It! - wdoekes On Oct. 27, 2014, 12:12 p.m., Corey Farrell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4115/ --- (Updated Oct. 27, 2014, 12:12 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24457 https://issues.asterisk.org/jira/browse/ASTERISK-24457 Repository: Asterisk Description --- fax_gateway_framehook leaks translated frames. Over 1.6mb worth of frames is leaked by tests/fax/sip/gateway_g711_t38 by one of the instances of Asterisk. Diffs - /branches/11/res/res_fax.c 426232 Diff: https://reviewboard.asterisk.org/r/4115/diff/ Testing --- Verified tests/fax/sip/gateway_g711_t38 no longer leaks on 11 and 13. Thanks, Corey Farrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4111: app_queue: ao2_iterator not destroyed, causing leak
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4111/#review13598 --- Ship it! Ship It! - rmudgett On Oct. 27, 2014, 12:53 a.m., Corey Farrell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4111/ --- (Updated Oct. 27, 2014, 12:53 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24454 https://issues.asterisk.org/jira/browse/ASTERISK-24454 Repository: Asterisk Description --- update_realtime_members doesn't release the iterator for q-members, causing a reference leak. Diffs - /branches/11/apps/app_queue.c 426232 Diff: https://reviewboard.asterisk.org/r/4111/diff/ Testing --- Testsuite against 13, compile test/visual inspection for 11. Thanks, Corey Farrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4110: manager: acl_change_sub leaks
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4110/#review13599 --- Ship it! Ship It! - Matt Jordan On Oct. 27, 2014, 12:38 a.m., Corey Farrell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4110/ --- (Updated Oct. 27, 2014, 12:38 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-24453 https://issues.asterisk.org/jira/browse/ASTERISK-24453 Repository: Asterisk Description --- Unsubscribe from acl_change_sub at shutdown. Diffs - /branches/12/main/manager.c 426232 Diff: https://reviewboard.asterisk.org/r/4110/diff/ Testing --- Ran testsuite against Asterisk 13, verified tests/manager/acl-login no longer leaks the reference. Visually inspected code and compiled for 12 to verify the issue applied to that version. Thanks, Corey Farrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4106: configure: Add autoconf check for libopus
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4106/#review13600 --- Ship it! I think this is fine to go into 13. - Matt Jordan On Oct. 22, 2014, 2:35 p.m., Sean Bright wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4106/ --- (Updated Oct. 22, 2014, 2:35 p.m.) Review request for Asterisk Developers. Repository: Asterisk Description --- Because opus transcoding support cannot be included in the standard Asterisk distribution, a few codec_opus implementations have popped up. To make it easier for people to drop in opus support in their own installations, this patch adds configure checks for libopus. I don't see why this wouldn't be safe for 13, but I will defer to the reviewers on that decision. Diffs - /trunk/makeopts.in 426095 /trunk/include/asterisk/autoconfig.h.in 426095 /trunk/configure.ac 426095 /trunk/configure UNKNOWN /trunk/build_tools/menuselect-deps.in 426095 Diff: https://reviewboard.asterisk.org/r/4106/diff/ Testing --- Thanks, Sean Bright -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 4106: configure: Add autoconf check for libopus
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4106/ --- (Updated Oct. 27, 2014, 12:54 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 426234 Repository: Asterisk Description --- Because opus transcoding support cannot be included in the standard Asterisk distribution, a few codec_opus implementations have popped up. To make it easier for people to drop in opus support in their own installations, this patch adds configure checks for libopus. I don't see why this wouldn't be safe for 13, but I will defer to the reviewers on that decision. Diffs - /trunk/makeopts.in 426095 /trunk/include/asterisk/autoconfig.h.in 426095 /trunk/configure.ac 426095 /trunk/configure UNKNOWN /trunk/build_tools/menuselect-deps.in 426095 Diff: https://reviewboard.asterisk.org/r/4106/diff/ Testing --- Thanks, Sean Bright -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] AppKonference 2.6
I have released an updated AppKonference that compiles with Asterisk 13. You can download the latest code from source forge: sourceforge.net/projects/appkonference That said Asterisk 13 doesn’t get that much attention because I use Asterisk 1.4 + some hacks. Here’s a link to my Asterisk 1.4 github repository: https://github.com/pjalbrecht/asterisk You don’t need these hacks to use the module, but you may find them useful.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] AstriDevCon 2014:Agenda item Deprecate AMI/AGI (Ben Klang)
The reason the dial plan can never be deprecated is because Asterisk wouldn’t be Asterisk without the dial plan. Sure, you could re-engineer Asterisk so that it would be “better for a small select group of users at the expense of the majority of community that use the product as designed for the purpose it was originally intended. However, you’re either very naive or delusional if you think the community is going to follow you down that path. Do you really believe the community is going simply chuck their dial plans and walk away from their investment in Asterisk? Not likely, dude. On Oct 24, 2014, at 11:48 AM, Jeffrey Ollie j...@ocjtech.us wrote: On Fri, Oct 24, 2014 at 10:09 AM, Paul Albrecht palbre...@glccom.com wrote: When Matt says deprecating the dial plan would be difficult and would take a long time it seems to me he’s being evasive and misleading. He doesn’t say it’s never going to happen and he doesn’t share whatever he thinks the Asterisk vision actually is which he should presumably be aware of since he is the Asterisk engineering manager. Why do you keep insisting that Digium promise to *never* deprecate dial plans? I don't think that's a promise that's really worth anything as there may be really good reasons in the future to do so. I think that you've gotten the best that you will get: they've said that there are no plans within Digium to deprecate the dial plan, and if there were plans, they'd give people a long time prepare before it actually happens. It's probably a good time to refresh your understanding of Digium's support policies: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Version 13 will be around until at least 2018, so you'll have *at least* that long to prepare for the switch, since version 13 is feature frozen so there's no way the dial plan would be removed from 13. And all of this talk of deprecating the dial plan isn't even coming from Digium. It's something that was suggested by a community member at the developer conference. I wasn't there so I don't know how seriously it was taken there, but it would have been impolite of everyone involved to just ignore it. -- Jeff Ollie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] AstriDevCon 2014:Agenda item Deprecate AMI/AGI (Ben Klang)
On Mon, Oct 27, 2014 at 2:04 PM, Paul Albrecht palbre...@glccom.com wrote: The reason the dial plan can never be deprecated is because Asterisk wouldn’t be Asterisk without the dial plan. Sure, you could re-engineer Asterisk so that it would be “better for a small select group of users at the expense of the majority of community that use the product as designed for the purpose it was originally intended. However, you’re either very naive or delusional if you think the community is going to follow you down that path. Do you really believe the community is going simply chuck their dial plans and walk away from their investment in Asterisk? Not likely, dude. My comment/question wasn't really about dial plans, per se. My question was about you insisting that Digium make such unqualified promises about the future of Asterisk. Even though Digium is a private company, I believe that they are still bound by U.S. laws regarding forward-looking statements[1]. So even if they wanted to (which I doubt), there's no way you're going to get the promise that you're looking for. [1] http://en.wikipedia.org/wiki/Forward-looking_statement -- Jeff Ollie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [Code Review] 4116: res_pjsip: incorrect qualify statistics after disabling for contact
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4116/ --- Review request for Asterisk Developers and Mark Michelson. Bugs: ASTERISK-24462 https://issues.asterisk.org/jira/browse/ASTERISK-24462 Repository: Asterisk Description --- When removing the qualify_frequency from an AoR or a contact the statistics shown when issuing pjsip show aors from the CLI are incorrect. This patch deletes the contact's status object from sorcery, disassociating it from the contact, if the qualify_freqency is removed from configuration. Diffs - branches/12/res/res_pjsip/pjsip_options.c 426251 Diff: https://reviewboard.asterisk.org/r/4116/diff/ Testing --- Using static and dynamic contacts and various combinations of adding, removing, and reloading the configuration for both AoR and contact level qualify_freqency options noted that the qualify statistics are now correctly reflected. Thanks, Kevin Harwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] SRTP in chan_motif.c
dwal...@fifo99.com wrote: On Mon, Oct 27, 2014 at 07:01:10PM -0300, Joshua Colp wrote: As there is no crypto support written it is not possible to negotiate it. Support would have to be added. All 3 variants are implemented. Non-Documented Google, Google, and XEP. So Google Voice's protocol is not documented? In the source documentation it broken down the naming of the different protocols. It wasn't clear to me which one was used for Google Voice.. Google Voice uses an undocumented version of Jingle which may or may not continue to exist in the future. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] SRTP in chan_motif.c
dwal...@fifo99.com wrote: On Mon, Oct 27, 2014 at 07:24:27PM -0300, Joshua Colp wrote: dwal...@fifo99.com wrote: On Mon, Oct 27, 2014 at 07:01:10PM -0300, Joshua Colp wrote: As there is no crypto support written it is not possible to negotiate it. Support would have to be added. All 3 variants are implemented. Non-Documented Google, Google, and XEP. So Google Voice's protocol is not documented? In the source documentation it broken down the naming of the different protocols. It wasn't clear to me which one was used for Google Voice.. Google Voice uses an undocumented version of Jingle which may or may not continue to exist in the future. Ok, where did you get the specification to write the chan_motif? Was it thru an NDA agreement, or reverse engineering ? It is based on the original code which I assume was reverse engineered. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
