search for: mmichelson

...| |--------------------+---------------------------------------------------| | Reported By | Russell Bryant <russell at digium.com> | | | | | | Mark Michelson <mmichelson at digium.com> | |--------------------+---------------------------------------------------| | Posted On | October 9, 2007 | |--------------------+---------------------------------------------------| | Last Updated On | Octo...

...| |--------------------+---------------------------------------------------| | Reported By | Russell Bryant <russell at digium.com> | | | | | | Mark Michelson <mmichelson at digium.com> | |--------------------+---------------------------------------------------| | Posted On | October 9, 2007 | |--------------------+---------------------------------------------------| | Last Updated On | Octo...

Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff..

...hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9300 (Brasil) ________________________________________ De: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] em Nome de Mark Michelson [mmichelson at digium.com] Enviado: sexta-feira, 10 de julho de 2015 15:14 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Can I use PJSIP_HEADER to read the SIP 183 message header? On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote: > Hi. > > The ASTER...

This bug report http://bugs.digium.com/print_bug_page.php?bug_id=12038 apparently has been fixed. I dont see anything on the page saying what released version of asterisk this is in. How can I tell that? jerry

...-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) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.7 Thank you for your continued supp...

...-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) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.7 Thank you for your continued supp...

...se 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 ba...

...se 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 ba...

Hello Asterisk community, We are having some problems with crashes in Asterisk, my asterisk versions are 1.4.24.1 and 1.4.23.2. I have found this: "~/work/asterisk-branch-1.4$ svn log -c 260345 ------------------------------------------------------------------------ r260345 | mmichelson | 2010-04-30 22:08:15 +0200 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it...

...eLog http://svn.asterisk.org/svn/asterisk/tags/1.6.1.5/ChangeLog The following list of issues were resolved with the participation of the community, and these releases would not have been possible without your help! * Fix SIP transport type issues. (closes issue #13865. Reported by st. Tested by mmichelson, Kristijan, vrban, jmacz, dvossel. Patched by: mmichelson, vrban, Kristijan) * Fix an issue where the 'h' extension may occasionally not fire when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. Fixed by Tilghman. * Fix MWI NOTIFY if Asterisk listens on a...

...eLog http://svn.asterisk.org/svn/asterisk/tags/1.6.1.5/ChangeLog The following list of issues were resolved with the participation of the community, and these releases would not have been possible without your help! * Fix SIP transport type issues. (closes issue #13865. Reported by st. Tested by mmichelson, Kristijan, vrban, jmacz, dvossel. Patched by: mmichelson, vrban, Kristijan) * Fix an issue where the 'h' extension may occasionally not fire when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. Fixed by Tilghman. * Fix MWI NOTIFY if Asterisk listens on a...

...MWI (Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman) * Fix possible segfault when logging (Closes issue #17331. Reported, patched by under. Patched by dvossel) * Fix memory hogging behavior of app_queue (Closes issue #17081. Reported by wliegel. Patched by mmichelson) * Allow type=user SIP endpoints to be loaded properly from realtime (Closes issue #16021. Reported, patched by Guggemand) Additionally, the following issue may be of interest: * Fix transcode_via_sln option with SIP calls and improve PLC usage (Review: https://reviewboard.asterisk.o...

...MWI (Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman) * Fix possible segfault when logging (Closes issue #17331. Reported, patched by under. Patched by dvossel) * Fix memory hogging behavior of app_queue (Closes issue #17081. Reported by wliegel. Patched by mmichelson) * Allow type=user SIP endpoints to be loaded properly from realtime (Closes issue #16021. Reported, patched by Guggemand) Additionally, the following issue may be of interest: * Fix transcode_via_sln option with SIP calls and improve PLC usage (Review: https://reviewboard.asterisk.o...

...Reported By Y Ateya Posted On 9 January, 2015 Last Updated On January 28, 2015 Advisory Contact Mark Michelson <mmichelson AT digium DOT com> CVE Name Pending Description Asterisk may be configured to only allow specific audio or video codecs to be used when communicating with a particular en...

...| |----------------------+-------------------------------------------------| | Last Updated On | December 9, 2008 | |----------------------+-------------------------------------------------| | Advisory Contact | Mark Michelson <mmichelson AT digium DOT com> | |----------------------+-------------------------------------------------| | CVE Name | | +------------------------------------------------------------------------+ +----------------------------------...

...bright) * Resolve crash in SLAtrunk when the specified trunk doesn't exist. (Reported in #asterisk-dev by philipp64. Patched by seanbright) * Update code to reflect that handle_speechset has 4 arguments. (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger, mmichelson) * Pass the PID of the Asterisk process, not the PID of the canary. (Closes issue #17065. Reported by globalnetinc. Patched by makoto. Tested by frawd, globalnetinc) * Resolve a deadlock in chan_local. (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2)...

...g 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 fo...

The Asterisk.org development team has released Asterisk versions 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, as well as Asterisk-Addons versions 1.6.0.1 and 1.6.1-rc2. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a fix for a regression introduced in Asterisk 1.2.30 and Asterisk 1.4.21.2 and has existed in the

...| |----------------------+-------------------------------------------------| | Last Updated On | December 9, 2008 | |----------------------+-------------------------------------------------| | Advisory Contact | Mark Michelson <mmichelson AT digium DOT com> | |----------------------+-------------------------------------------------| | CVE Name | | +------------------------------------------------------------------------+ +----------------------------------...