478 Commits (02a124eda58808fc50418cbb2217772e47e440e4)

Author	SHA1	Message	Date
Joshua Colp	17be8b9c20	Merge "install_prereq: Update repositories before install on Debian systems"	10 years ago
Rodrigo Ramírez Norambuena	88f3dbaec9	install_prereq: Update repositories before install on Debian systems ... When to install packages the indexed local is more old of the version of software on the repository they have been upgraded by security update then get the package will give 404 not found. The patch prevent by update local index to repository for aptitude before install. ASTERISK-25495 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I645959e553aac542805ced394cac2dca964051fa	10 years ago
Kevin Harwell	691c0e0b31	res_pjsip_outbound_registration: registration stops due to fatal 4xx response ... During outbound registration it is possible to receive a fatal (any permanent/ non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due to a problem with the registrar itself. Upon receiving the failure response Asterisk terminates outbound registration for the given endpoint. This patch adds an option, 'fatal_retry_interval', that when set continues outbound registration at the given interval up to 'max_retries' upon receiving a fatal response. ASTERISK-25485 #close Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2	10 years ago
Matt Jordan	b9bd249a85	contrib/scripts/autosupport: Update for Asterisk 13 ... This patch adds some minor tweaks for autosupport to update it for Asterisk 13. This includes: * Finally removing most references to Zaptel * Adding support for some additional 'core' commands, and fixing nomenclature that generally hasn't been used for some time * Adding some PJSIP/SIP commands to gather endpoints/peers and active channels Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1 (cherry picked from commit 9fc9777fa3)	10 years ago
Joshua Colp	96c26f497a	Merge "Scripts: check file versions of Asterisk and dependencies"	10 years ago
Scott Griepentrog	d6472d96b3	Scripts: check file versions of Asterisk and dependencies ... To help in diagnosing mismatched modules and libraries, this script scans for version, repository, and source information and reports what is found. ASTERISK-25376 #close Reported by: Ashley Sanders Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6	10 years ago
Mark Michelson	993ae9a669	res_pjsip: Change default from user value. ... When Asterisk sends an outbound SIP request, if there is no direct reason to place a specific value for the username in the From header, Asterisk would generate a UUID. For example, this would happen when sending outbound OPTIONS requests when qualifying or when sending outbound INVITE requests when originating (if no explicit caller ID were provided). The issue is that some SIP providers reject these sorts of requests with a "Name too long" error response. This patch aims to fix this by changing the default outbound username in From headers to "asterisk". This value can be overridden by changing the default_from_user option in the global options if desired. ASTERISK-25377 #close Reported by Mark Michelson Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190	10 years ago
Scott Griepentrog	53e2a6a829	contrib: script install_prereq should install sqlite3 ... Asterisk needs the sqlite 3 library, which is package sqlite-devel in CentOS. By adding this package to the script, a problem with configure failing is resolved. ASTERISK-25331 #close Reported by: Kevin Harwell Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec	10 years ago
Joshua Colp	309dd2a409	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. ... This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	10 years ago
Mark Michelson	2b42264e66	res_pjsip: Add rtp_keepalive endpoint option. ... This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d	10 years ago
Kevin Harwell	93ac45d3bd	res_pjsip: Add option to force G.726 to be treated as AAL2 packed. ... Some phones send g.726 audio packed for AAL2, which differs from what is recommended by RFC 3351. If Asterisk receives audio formatted as such when negotiating g.726 then it sounds a bit distorted. Added an option to res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 AAL2 packed. ASTERISK-25158 #close Reported by: Steve Pitts Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615	11 years ago
Rodrigo Ramírez Norambuena	6737ded058	install_prereq: Check if is installed aptitude otherwise to install. ... If in Debian or system based, dont have aptitude installed the script do nothing. This patch checked if aptitude installed, if not installed. Also, if execute script with all packages installed yet, the script not show nothing and return exit 1 because the command 'grep' get nothing from pipe from 'awk'. ASTERISK-25113 #close Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> Change-Id: Iebdff55805d3917166e5e08e0a1e2176f36ff27f	11 years ago
Corey Farrell	478fb4a388	MALLOC_DEBUG: Replace WRAP_LIBC_MALLOC with ASTMM_LIBC. ... There are 3 ways that calls directly to standard allocator functions can be dealt with: 1. Block their use, cause them to generate an error. This is the default. 2. Replace them with the Asterisk equivalent function calls. 3. Leave them alone. This change allows one of these 3 options to be selected by any source. The source just needs to define ASTMM_LIBC to ASTMM_BLOCK, ASTMM_REDIRECT, or ASTMM_IGNORE to use option 1, 2 or 3 respectively. Normally ASTMM_BLOCK is the correct option, so it is default when ASTMM_LIBC is not defined. In some cases when building 3rd party code it is desirable to have it use Asterisk functions, without changing the whole source - ASTMM_REDIRECT accomplishes this. When using 3rd party libraries sometimes a static inline function will make use of malloc or free. In these cases it may be unsafe to replace the allocator in the header, as it's possible the memory could be freed by the library using standard allocators. For those cases ASTMM_IGNORE is needed. Change-Id: I8afef4bc7f3b93914263ae27d3a5858b69663fc7	11 years ago
George Joseph	cf637f2510	doc: Make progdocs play nice with git ... Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in Changed /Makefile to copy asterisk-ng-doxygen.in to asterisk-ng-doxygen then modify it with version instead of modifying asterisk-ng-doxygen directly. Updated clean targets as well. Updated /.gitignore and doc/.gitignore. Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622	11 years ago
Ivan Poddubny	b885f719bf	contrib/editors: Fix vim syntax highlighting of comments in config files ... * Added a lookbehind to one-line comment matcher to skip escaped semicolons. * Added support for block comments. Change-Id: Id17dfaeda8ed4be572e8107a0c010066584aaee7	11 years ago
Matt Jordan	75c0aa6979	contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update ... The upgrade script for auto DTMF mode (31cd4f4891ec) added in 88b0fa7755 failed to add ENUM support for Postgres databases. This requires a specific import from the sqlalchemy.dialects.postgresql package. This patch corrects this error, which allows for Postgres update scripts to be generated. ASTERISK-24706 Change-Id: I4742ac8efa533cd6f18e0bdd907b339a9aedf015	11 years ago
Joshua Colp	7c18edaa10	Merge "Example script for scan-build (the llvm static analyzer)"	11 years ago
Corey Farrell	5c1d07baf0	Astobj2: Allow reference debugging to be enabled/disabled by config. ... * The REF_DEBUG compiler flag no longer has any effect on code that uses Astobj2. It is used to determine if reference debugging is enabled by default. Reference debugging can be enabled or disabled in asterisk.conf. * Caller information is provided in logger errors for ao2 bad magic numbers. * Optimizes AO2 by merging internal functions with the public counterpart. This was possible now that we no longer require a dual ABI. ASTERISK-24974 #close Reported by: Corey Farrell Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1	11 years ago
Diederik de Groot	1e74793061	Example script for scan-build (the llvm static analyzer) ... - Added Pre-amble (Options / Flags / Usage Example / GNU License) - Extended Configurability - Made Executable ASTERISK-24917 Change-Id: I70405fe54e4be7dbfbcb62e291690069b88617a8	11 years ago
George Joseph	c6ed681638	res_pjsip: Add global option to limit the maximum time for initial qualifies ... Currently when Asterisk starts initial qualifies of contacts are spread out randomly between 0 and qualify_timeout to prevent network and system overload. If a contact's qualify_frequency is 5 minutes however, that contact may be unavailable to accept calls for the entire 5 minutes after startup. So while staggering the initial qualifies is a good idea, basing the time on qualify_timeout could leave contacts unavailable for too long. This patch adds a new global parameter "max_initial_qualify_time" that sets the maximum time for the initial qualifies. This way you could make sure that all your contacts are initialy, randomly qualified within say 30 seconds but still have the contact's ongoing qualifies at a 5 minute interval. If max_initial_qualify_time is > 0, the formula is initial_interval = min(max_initial_interval, qualify_timeout * random(). If not set, qualify_timeout is used. The default is "0" (disabled). ASTERISK-24863 #close Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 Tested-by: George Joseph <george.joseph@fairview5.com>	11 years ago
George Joseph	51886c68dc	pjsip_options: Add qualify_timeout processing and eventing ... This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html The basic issues are that changes in contact status don't cause events to be emitted for the associated endpoint. Only dynamic contact add/delete actions update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds which is a long time. This patch makes use of the new transaction timeout feature in r4585 and provides the following capabilities... 1. A new aor/contact variable 'qualify_timeout' has been added that allows the user to specify the maximum time in milliseconds to wait for a response to an OPTIONS message. The default is 3000ms. When the timer expires, the contact is marked unavailable. 2. Contact status changes are now propagated up to the endpoint as follows... When any contact is 'Available', the endpoint is marked as 'Reachable'. When all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The existing endpoint events are generated appropriately. ASTERISK-24863 #close Change-Id: Id0ce0528e58014da1324856ea537e7765466044a Tested-by: Dmitriy Serov Tested-by: George Joseph <george.joseph@fairview5.com>	11 years ago
Matthew Jordan	8bae18ab93	res_pjsip: Add an 'auto' option for DTMF Mode ... This patch adds support for automatically detecting the type of DTMF that a PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto', the channel created for an endpoint will attempt to determine if RFC 4733 DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type for the channel will be set to inband. Review: https://reviewboard.asterisk.org/r/4438 ASTERISK-24706 #close Reported by: yaron nahum patches: yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676) ........ Merged revisions 434637 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	cb1c639817	Add missing file. ASTERISK-24781 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	79a81fed59	alemebic scripts: endpoint identifier order option ... The script was added in 13, but when committed to trunk it caused a branch to occur due to some trunk only alemebic changes. This fixes it so that the new 'add_pjsip_endpoint_identifier_order script points to the correct down revision. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	aef7278af6	res_pjsip: Allow configuration of endpoint identifier query order ... This patch fixes previously reverted code that caused binary incompatibility problems with some modules. And like the original patch it makes sure that no matter what order the endpoint identifier modules were loaded, priority is given based on the ones specified in the new global 'endpoint_identifier_order' option. ASTERISK-24840 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4489/ ........ Merged revisions 433028 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	d42c6adb1a	Revert - res_pjsip: Allow configuration of endpoint identifier query order ... Due to a break in binary compatibility with some other modules these changes are being reverted until the issue can be resolved. ASTERISK-24840 Reported by: Mark Michelson ........ Merged revisions 432868 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	1ce529d30e	res_pjsip: allow configuration of endpoint identifier query order ... It's possible to have a scenario that will create a conflict between endpoint identifiers. For instance an incoming call could be identified by two different endpoint identifiers and the one chosen depended upon which identifier module loaded first. This of course causes problems when, for example, the incoming call is expected to be identified by username, but instead is identified by ip. This patch adds a new 'global' option to res_pjsip called 'endpoint_identifier_order'. It is a comma separated list of endpoint identifier names that specifies the order by which identifiers are processed and checked. ASTERISK-24840 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4455/ ........ Merged revisions 432638 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	55709bc1f7	install_prereq: Tweak flags when configuring pjproject. ... This change does two things: 1. Disables debugging so assertions which can return an error do, instead of asserting. 2. Enables IPv6 support. ASTERISK-24632 #close Reported by: Rusty Newton ........ Merged revisions 431843 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	49cbfa7de6	Fix typo's (retrieve, specified, address). ... ........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430998 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	14b8e03dad	contrib/scripts/install_prereq: Don't install 32-bit packages on 64-bit hosts ... On Debian based systems, the install_prereq tool uses a search command on Debian that results in selecting both 64-bit and 32-bit packages. Besides the waste of disk space, this can actually cause aptitude use 100% of memory on a VM with 1GB of RAM as it tried to work out all of the 32-bit package dependencies. This patch filters out the 32-bit packages on a 64-bit machine, and leaves 32-bit machines alone. ASTERISK-24048 #close Reported by: Ben Klang Tested by: Ben Klang, Matt Jordan patches: install_prereq_64-bit_compat.patch uploaded by Ben Klang (License 5876) ........ Merged revisions 430798 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430799 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	fba836cc02	sip_to_pjsip: improve ability to parse input files ... General improvements to SIP to PJSIP conversion utility: 1) track default section of input file to allow parsing an include file that doesn't specify a [section] 2) informatively handle case of assignment without [section] 3) correctly handle getting sections from included files - [section]'s are inherited by included file 4) provide null string as default transport bind ip 5) gracefully handle missing portions of registration string 6) denote steps of operation during conversion and confirm top level files as a convenience ASTERISK-24474 #close Review: https://reviewboard.asterisk.org/r/4280/ Reported by: John Kiniston ........ Merged revisions 430469 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	7f8b7ace72	res_pjsip_sdp_rtp: Add support for optimistic SRTP. ... Optimistic SRTP is the ability to enable SRTP but not have it be a fatal requirement. If SRTP can be used it will be, if not it won't be. This gives you a better chance of using it without having your sessions fail when it can't be. Encrypt all the things! Review: https://reviewboard.asterisk.org/r/3992/ ........ Merged revisions 428222 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	b2e766a6b7	alembic: Fix alembic migration for 'moh_passthrough' option in res_pjsip. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ac091d4184	chan_pjsip: Add support for passing hold and unhold requests through. ... This change adds an option, moh_passthrough, that when enabled will pass hold and unhold requests through using a SIP re-invite. When placing on hold a re-invite with sendonly will be sent and when taking off hold a re-invite with sendrecv will be sent. This allows remote servers to handle the musiconhold instead of the local Asterisk instance being responsible. Review: https://reviewboard.asterisk.org/r/4103/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	bf684b63a3	REF_DEBUG: Install refcounter.py to $(ASTDATADIR)/scripts ... This change ensures refcounter.py is installed to a place where it can be found by the Asterisk testsuite if REF_DEBUG is enabled. ASTERISK-24432 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4094/ ........ Merged revisions 426830 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426831 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426832 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426833 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	7144c739e9	res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable. ... This change adds a configuration option which adds a 'user=phone' parameter if the user portion of the request URI or the From URI is determined to be a number. Review: https://reviewboard.asterisk.org/r/4073/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	d0255c4a46	safe_asterisk: Don't automatically exceed MAXFILES value of 2^20. ... On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can exceed the per-process file limit of 2^20. This patch ensures the value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626) ........ Merged revisions 424875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424878 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424879 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424880 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	b15cd42b5b	Alembic: Add enumerator value to sippeers -> directmedia - 'outgoing' ... The 'outgoing' value was left off of the enumerator when first creating the column. This patch adds it, and should gracefully upgrade keeping the existing data in tact. ASTERISK-23781 #close Reported by: Stephen More Review: https://reviewboard.asterisk.org/r/4013/ ........ Merged revisions 424372 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424373 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	b56dfb78c5	autosupport: Fix bashism. ... '==' is bashism (bashspecific, fails when dash is /bin/sh). Anyway, a 'case' works better there. Originally committed in r375059 and r375060 on 2012-10-16 21:13:08. ASTERISK-20567 #close Reported by: Tzafrir Cohen ........ Merged revisions 424117 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424125 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424126 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424127 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	3c1804eb0d	format_mp3: Made the get script conditionally apply patch if not already there. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	37179a2b1f	core: Don't allow free to mean ast_free (and malloc, etc..). ... This gets rid of most old libc free/malloc/realloc and replaces them with ast_free and friends. When compiling with MALLOC_DEBUG you'll notice it when you're mistakenly using one of the libc variants. For the legacy cases you can define WRAP_LIBC_MALLOC before including asterisk.h. Even better would be if the errors were also enabled when compiling without MALLOC_DEBUG, but that's a slightly more invasive header file change. Those compiling addons/format_mp3 will need to rerun ./contrib/scripts/get_mp3_source.sh. ASTERISK-24348 #related Review: https://reviewboard.asterisk.org/r/4015/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a7add3a257	astobj2.c/refcounter.py: Fix to deal with invalid object refs. ... * Make astob2 REF_DEBUG output an invalid object line when an invalid ao2 object ref/unref is attempted. This is similar to the constructor/destructor lines. * Fixed refcounter.py to handle skewed objects that have constructor/destructor states. * Made refcounter.py highlight the invalid ao2 object refs by putting them in their own section of the processed output file. * Made refcounter.py highlight unreffing an object by more than one that results in a negative ref count and the object being destroyed. The abnormally destroyed object is reported in the invalid and finalized object sections of the output. Review: https://reviewboard.asterisk.org/r/3971/ ........ Merged revisions 423349 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423400 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423416 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423418 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423422 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	77834b72d3	contrib: Fix verifyi typo in alembic DB script ps_transport table. ... Reported by: Zogot (on IRC) Patches: tmp.diff uploaded by Zogot, cleaned up by me. ........ Merged revisions 423128 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423129 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	add46fd27c	app_queue: Add RealTime support for queue rules ... This patch gives the optional ability to keep queue rules in RealTime. It is important to note that with this patch: (a) Queue rules in RealTime are only examined on module load/reload (b) Queue rules are loaded both from the queuerules.conf file as well as the RealTime backend To inform app_queue to examine RealTime for queue rules, a new setting has been added to queuerules.conf's general section "realtime_rules". RealTime queue rules will only be used when this setting is set to "yes". The schema for the database table supports a rule_name, time, min_penalty, and max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or '+' literal is provided. Otherwise, the penalties are treated as constants. For example: rule_name, time, min_penalty, max_penalty 'default', '10', '20', '30' 'test2', '20', '30', '55' 'test2', '25', '-11', '+1111' 'test2', '400', '112', '333' 'test3', '0', '4564', '46546' 'test_rule', '40', '15', '50' which would result in : Rule: default - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule: test2 - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564 Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the queue rules will be always reloaded on a module reload, even if the underlying file did not change. With the option disabled, the rules will only be reloaded if the file was modified. Review: https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close Reported by: Michael K patches: app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621) ........ Merged revisions 420624 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	8bdf374c4a	Fix alembic script to work properly in offline mode. ... When run in offline mode, this would attempt to check the database for the presence of a type it was going to try to create. I now check the context to see if we're running in offline mode and change a parameter accordingly. ........ Merged revisions 407567 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	5273a14839	Add alembic script that adds contact user_agent and endpoint message_context. ... ........ Merged revisions 411514 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a6612e80aa	alembic: Adjust sippeers, queue_members, and voicemail_messages tables. ... * Increased the sippeers useragent max string size to 255. * Changed the queue_members uniqueid to an auto incremented integer instead of a string. * Increased the voicemail_messages BLOB size to LONGBLOB on mysql. * Fixed the add_tables_for_pjsip config change version downgrade actions to drop a table it created. * Adjusted the sample alembic.ini files cdr.ini.sample, config.ini.sample, and voicemail.ini.sample to give a mysql and postgres sqlalchemy.url lines. ASTERISK-23847 #close Reported by: Stephen More ASTERISK-23825 #close Reported by: Stephen More ASTERISK-23909 #close Reported by: Stephen More Review: https://reviewboard.asterisk.org/r/3870/ ........ Merged revisions 420211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fd94fea599	res_pjsip: Support setting a default accountcode on endpoints ... Most channel drivers let you specify a default accountcode to be set on channels associated with a particular peer/endpoint/object. Prior to this patch, chan_pjsip/res_pjsip did not support such a setting. This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'. When a channel is created that is associated with an endpoint with this value set, the channel will automatically have its accountcode property set to the value configured for the endpoint. Review: https://reviewboard.asterisk.org/r/3724/ ASTERISK-24000 #close Reported by: Matt Jordan ........ Merged revisions 418756 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	fcdc4ad0bf	astobj2: work around REF_DEBUG race which causes out of order log entries ... * Update refcounter.py to use delta's to track the current reference count. * Use result from internal_ao2_ref to write old_refcount to refs_log. Review: https://reviewboard.asterisk.org/r/3756/ ........ Merged revisions 418504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 418505 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 418506 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	da469fd9f6	dahdi_span_config_hook: automatically register new dahdi channels ... Install a hook script for DAHDI to register new spans with Asterisk automatically by running: asterisk -rx 'dahdi create channel FIRST LAST' Review: https://reviewboard.asterisk.org/r/3157/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago