search for: cmaj

...a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis) * ASTERISK-29021 - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions (Reported by cmaj) * ASTERISK-28927 - Asterisk crash in music on hold (Reported by David Cunningham) * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address) (Reported by Michael Neuhauser) * ASTER...

...a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis) * ASTERISK-29021 - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions (Reported by cmaj) * ASTERISK-28927 - Asterisk crash in music on hold (Reported by David Cunningham) * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address) (Reported by Michael Neuhauser) * ASTER...

...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 Geo...

...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 Geo...

...a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis) * ASTERISK-29021 - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions (Reported by cmaj) * ASTERISK-28927 - Asterisk crash in music on hold (Reported by David Cunningham) * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address) (Reported by Michael Neuhauser) * ASTER...

...a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis) * ASTERISK-29021 - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions (Reported by cmaj) * ASTERISK-28927 - Asterisk crash in music on hold (Reported by David Cunningham) * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address) (Reported by Michael Neuhauser) * ASTER...

...ed 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/se...

...ed 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/se...

...calls, no ringing indication is played (Reported by Jaco Kroon) * ASTERISK-30424 - pjproject_bundled: cross-compilation broken when ssl autodetected (Reported by Nick French) * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when multi-homed (Reported by cmaj) * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by Ross Beer) * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember (Reported by Sean Bright) * ASTERISK-30406 - pbx_ael: Global variables are not expanded. (Reported by Sean Bright)...

...calls, no ringing indication is played (Reported by Jaco Kroon) * ASTERISK-30424 - pjproject_bundled: cross-compilation broken when ssl autodetected (Reported by Nick French) * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when multi-homed (Reported by cmaj) * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by Ross Beer) * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember (Reported by Sean Bright) * ASTERISK-30406 - pbx_ael: Global variables are not expanded. (Reported by Sean Bright)...

...calls, no ringing indication is played (Reported by Jaco Kroon) * ASTERISK-30424 - pjproject_bundled: cross-compilation broken when ssl autodetected (Reported by Nick French) * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when multi-homed (Reported by cmaj) * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by Ross Beer) * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember (Reported by Sean Bright) * ASTERISK-29604 - ari: Segfault with lots of calls (Reported by Danila Evgrafov) *...

...calls, no ringing indication is played (Reported by Jaco Kroon) * ASTERISK-30424 - pjproject_bundled: cross-compilation broken when ssl autodetected (Reported by Nick French) * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when multi-homed (Reported by cmaj) * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by Ross Beer) * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember (Reported by Sean Bright) * ASTERISK-29604 - ari: Segfault with lots of calls (Reported by Danila Evgrafov) *...

I have setup SAMBA to authenticate to a 2003 (probably SP1) domain and I am having two very distinct problems. I have searched the archives, but each time I see this question is ask I never see any answers and I need an answer or I am going to have to shut this off. I work in the Kentucky school systems and we have one AD forest containing one domain for each county in the state. That totals up

...Improvements made in this release: ----------------------------------- * ASTERISK-28710 - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) * ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) * ASTERISK-28673 - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) * ASTERISK-28658 - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog...

...Improvements made in this release: ----------------------------------- * ASTERISK-28710 - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) * ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) * ASTERISK-28673 - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) * ASTERISK-28658 - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog...

...Improvements made in this release: ----------------------------------- * ASTERISK-28710 - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) * ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) * ASTERISK-28673 - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) * ASTERISK-28658 - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog...

...Improvements made in this release: ----------------------------------- * ASTERISK-28710 - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) * ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) * ASTERISK-28673 - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) * ASTERISK-28658 - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog...

...Improvements made in this release: ----------------------------------- * ASTERISK-28710 - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) * ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) * ASTERISK-28673 - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) * ASTERISK-28658 - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog...

...Improvements made in this release: ----------------------------------- * ASTERISK-28710 - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) * ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) * ASTERISK-28673 - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) * ASTERISK-28658 - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog...

...on: Maximum retries reached (Reported by Daniel) * ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam Banks) * ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column (Reported by cmaj) * ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup. (Reported by Michael) * ASTERISK-28542 - [patch] add the ability for asterisk to generate on-hold re-invites (Reported by Torrey Searle)...