24715 Commits (3ff403fc31ecbef308928671a29c3f8a0a69ebf2)

 

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Author	SHA1	Message	Date
David M. Lee	3ff403fc31	My doc correction in r400842 had a silly bug. ... Because I added a wiki_description to models and not their properties, the rendered wiki page had the model description instead of the property descriptions, which looks very silly indeed. (closes issue ASTERISK-22705) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	02d57251b4	chan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" output. ... * Remember the swgain setting from CLI "dahdi set swgain" command so the CLI "dahdi show channel" output will reflect the current setting. * Updated CLI "dahdi set hwgain" and "dahdi set swgain" documentation. (issue ASTERISK-22429) Reported by: Jaco Kroon Patches: jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 400907 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400909 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	84adf58988	chan_sip: Do not increment the SDP version between 183 and 200 responses. ... Bumping the SDP version number can cause interoperability problems since receivers of the responses will expect that a 200 SDP will be identical to a previous 183 SDP. (closes issue ASTERISK-21204) reported by NITESH BANSAL Patches: dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418) ........ Merged revisions 400906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400908 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	cf37f8d4c4	pjsip outbound registration: Log message says received a 408 when we didn't ... If the server didn't exist that we are trying to register to the log message would say that a 408 was received from that server when in reality one wasn't. Added log messages stating no response was received if the response does not exist. (closes issue ASTERISK-22554) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2893/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	02e02739ce	Remove duplicate module info block ... The module info block was repeated twice. Once is sufficient. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e94f240a88	Fix a race condition in res_pjsip_session with rapidly terminating the session. ... The INVITE session state callback wrongly assumes that a session will always exist, but when rapidly terminating the session this assumption goes out the window. As all handler code for the INVITE session state callback requires the session it will now just exit immediately if no session exists. (closes issue ASTERISK-22668) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	07204b45c3	Fix realm comparison for outbound auth ... When generating the list of authentication credentials to pass to PJSIP, Asterisk was using the raw pointer of a pj_str_t which is not always NULL-terminated. This sometimes resulted in incorrect text for the realm and a failure to match the realm for authentication purposes which was causing the outbound nominal auth pjsip basic call test to bounce. This now uses the pj_str_t that contains the realm instead of generating a new one. Thanks to John Bigelow for helping to narrow this down. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3c20ac4f61	channel.h: whitespace changes. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	eba9eb5e1b	Softmix: Fix crash when switching from softmix to another bridge technology. ... The crash is caused by a race condition when switching between native RTP and softmix bridging technologies. In this situation, the bridging technology is switched from native RTP to softmix, and then back to native RTP fast enough that the softmix private data gets destroyed before the softmix mixing thread gets started. Thanks to Kinsey Moore for the crash analysis. * Fix race condition when starting the softmix mixing thread and switching to another bridge technology. (closes issue ASTERISK-22678) Reported by: John Bigelow Patches: jira_asterisk_22678_v12.patch (license #5621) patch uploaded by rmudgett Tested by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400849 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	1af791acb0	Fix a stupid copy/paste error in ARI docs. ... Patches: ari-doc-patch.txt uploaded by jbigelow (license 5091) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	945108058c	Updated /play resource docs. The playback of http: resources isn't implemented... yet ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	76be693d83	Correct some ARI wiki rendering errors ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	3bc6dd4f7b	Perform validation of permanent contacts on AORs in res_pjsip. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	930351f543	Fix an assertion in res_pjsip when specifying an invalid outbound proxy. ... This change fixes two issues when setting an outbound proxy: 1. The outbound proxy URI was not parsed and validated during configuration. 2. If an outgoing dialog was created and the outbound proxy could not be set an assertion would occur because the usage count on the dialog was not decremented. The documentation has also been updated to specify that a full URI must be specified for the outbound proxy. (closes issue ASTERISK-22672) Reported by: Antti Yrjola git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7372b09ab9	Use 'z' as the format specifier for size_t ... Using 'lu' will produce a compiler warning for some versions of gcc and on some architectures. 'z' should be portable as a format specifier for size_t. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f440c14a37	Add PJSIP_HEADER function for manipulation of SIP headers in the PJSIP stack ... This patch adds support to the PJSIP stack in Asterisk for SIP header manipulation. Note that this is analagous to SIPAddHeader/SIPRemoveHeader. For PJSIP_HEADER, an incoming supplemental session callback is registered that takes the pjsip_hdrs from the incoming session and stores them in a linked list in the session datastore. Calls to PJSIP_HEADER traverse over the list and return the nth matching header where 'n' is the 'number' argument to the function. When adding a header, the first call creates a datastore and linked list and adds the datastore to the session. The header is then created as a pjsip_hdr and added to the list. An outgoing supplemental session callback then traverses the list and adds the headers to the outgoing pjsip_msg. When removing a header, the list created with PJSIP_HEADER(add,...) is traversed and all matching entries are removed. (closes issue ASTERISK-22498) Reported by: George Joseph patch: res_pjsip_header_funcs_v1.patch uploaded by george.joseph (License 6322) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	b919e0a72c	Add warning when compiling with iODBC support ... When running configure, libiodbc2 development headers will fulfill the requirement for ODBC development headers, but will not function properly. This adds a warning when libiodbc2 development headers are detected instead of unixodbc development headers. (closes issue ASTERISK-22459) Reported by: Patrick Maille Tested by: Walter Doekes Patches: issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes (License 5674) ........ Merged revisions 400767 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400768 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	3e1485af09	app_agent_pool: Fix AMI/CLI AgentLogoff soft preventing agents from logging back in. ... * Clear the deferred_logoff flag when an agent logs in. (closes issue ASTERISK-22669) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400754 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	7f60f26ed2	Switch from using pjsip_strerror to pj_strerror. ... pjsip_strerror is only aware of PJSIP-specific error codes. pj_strerror() is aware of all PJProject error codes and OS-specific error codes. This specifically fixes an oft-seen error in transport configuration code where EADDRINUSE would result in "Unknown PJSIP error 120098" instead of a useful message. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	25f54f1fe2	app_confbridge: Can now set the language used for announcements to the conference. ... ConfBridge now has the ability to set the language of announcements to the conference. The language can be set on a bridge profile in confbridge.conf or by the dialplan function CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983) Reported by: Jonathan White Patches: M19983_rev2.diff (license #5138) patch uploaded by junky (modified) Tested by: rmudgett ........ Merged revisions 400741 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400742 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ea5a8334c6	app_confbridge: Fix duplicate default_user profile. ... * Fixed looking in the wrong profiles container to see if the default_user profile is already created in verify_default_profiles(). The bridge profile container is never going to hold user profiles. :) ........ Merged revisions 400723 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	b671dcde04	Fix func_config list entry allocation ... The AST_CONFIG dialplan function defined in func_config.c allocates its config file list entries using ast_malloc. List entry allocations destined for use with Asterisk's linked list API must be ast_calloc()d or otherwise initialized so that list pointers are set to NULL. These uses of ast_malloc have been replaced by ast_calloc to prevent dereferencing of uninitialized pointer values when traversing the list. (closes issue ASTERISK-22483) Reported by: Brian Scott ........ Merged revisions 400694 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400697 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	b04d30f1c7	Fix STUN crash when using IPv6 any address ... Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4 candidates are also added. (closes issue ASTERISK-21917) Reported by: Torrey Searle Patches: 0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334) ........ Merged revisions 400681 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	ce3d908fca	Push CLI qualify into the threadpool. ... If you run Asterisk in the background and then connect to it through a separate console, the thread that runs CLI commands is not registered with PJLIB. Thus PJLIB does not like it when you attempt to send OPTIONS requests from that thread. So now we push the task into the threadpool, which we know to be registered with PJLIB. Thanks to Antti Yrjola for reporting this. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e57deaec33	Make app_queue and res_agi independent of AMI being enabled. ... The https://reviewboard.asterisk.org/r/2888/ review changes manager to not subscribe to stasis when it is disabled for performance reasons. When manager is disabled app_queue and res_agi decline to load and fail to clean up what they have already allocated. * Made app_queue and res_agi clean up allocated resources when they decline to load. * Made app_queue and res_agi use their own subscriptions to the stasis topics instead of borrowing manager's message router structure inappropriately. (closes issue ASTERISK-22604) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/2902/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1d72d481a7	Miscellaneous stand alone comment cleanups. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	74320e6955	app_queue: Fix Queuelog EXITWITHKEY only logging two of four fields ... Commit r62462 added two extra fields for logging "the original position the caller entered the queue at, and the amount of time the caller was waiting in the queue." But when r75969 was merged from 1.4 into trunk (r75977), these two fields disappeared. Those two extra fields were not logged in 1.4 and when the patch was merged, those fields went away. Therefore, this is a regression and was caught by the reporter because he was reading the awesome "Asterisk: The Definitive Guide" book. (closes issue ASTERISK-22197) Reported by: Dalius M. Tested by: Dalius M. Patches: asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2901/ ........ Merged revisions 400622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400623 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e848dbab4f	chan_iax2: Fix compile error. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	130fd15c24	Add IPv6 Support To chan_iax2 ... This patch adds IPv6 support to chan_iax2. Yay! (closes issue ASTERISK-22025) Patches: iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2660/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	cd2ddccaf2	Added missing file from r400522 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	c794605ea5	chan_pjsip: Make logger togglable without loading/unloading ... This patch makes the res_pjsip_logger do a few things... First, it will be built and installed by default now, so end users won't need to enable it in menuselect. Second, while it is loaded, it no longer will immediately issue log messages. Upon loading, it is in the disabled state and must be turned on with the new CLI command. The CLI command 'pjsip set logger <on/off/host> has been added and can be used to do the following: pjsip set logger on: Enables logger for all PJSIP traffic pjsip set logger off: Disables logger for all PJSIP traffic pjsip set logger host <host>: Enables logger for the specific host Review: https://reviewboard.asterisk.org/r/2900/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	66137de7e8	chan_pjsip: Add alembic scripts for generating db tables for PJSIP ... Also updates sample configurations for sorcery and extconfig to demonstrate how to use databases created by that alembic script. (closes issue ASTERISK-22133) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2892/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7fc567bd76	ARI: Add subscription support ... This patch adds an /applications API to ARI, allowing explicit management of Stasis applications. * GET /applications - list current applications * GET /applications/{applicationName} - get details of a specific application * POST /applications/{applicationName}/subscription - explicitly subscribe to a channel, bridge or endpoint * DELETE /applications/{applicationName}/subscription - explicitly unsubscribe from a channel, bridge or endpoint Subscriptions work by a reference counting mechanism: if you subscript to an event source X number of times, you must unsubscribe X number of times to stop receiveing events for that event source. Review: https://reviewboard.asterisk.org/r/2862 (issue ASTERISK-22451) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	c1e76f6ccb	Enclose the To URI and update its user portion if a request user has been specified. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	bc81a9000f	Replace the connection address at the SDP level if altering the SDP with the external media address. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400510 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	096ce6c5b7	Corrected response class for stopPlayback ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	624dbb74a5	chan_sip: Don't ignore expires value in contact header if it lacks semicolon ... (closes issue ASTERISK-22574) Reported by: Filip Jenicek Patches: chan_sip_expires.patch uploaded by Filip Jenicek (license 6277) ........ Merged revisions 400469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400470 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	19df630819	Remove publication of a channel snapshot when the technology is set ... This patch removes said publication for a few reasons: (1) It is unnecessary. Association of the channel technology with a specific channel is an implementation detail that should be assumed to "just happen", and consumers of Stasis don't need to be informed about it. (2) Publication of said message can now cause crashes, as the actual creation of a channel in normal locations now stages its messages. As a result, things that create dummy channels (such as the SIP RTP QOS unit test) and associate them with a channel technology were now crashing, as the channel itself was not known by Stasis. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	7cfd95ac44	When serializing CDR variables (like for "core show channels") don't output an error if CDRs aren't enabled. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	1f9ac47811	Fix security events for AMI invalid password ... In r337595, additional security events were added for chan_sip authentication failures. The new IEs added to the existing invalid password event were defined as required IEs, but existing users of the event did not set the new IEs and could not since they didn't apply to existing uses. They are now marked as optional IEs. (closes issue ASTERISK-22578) Reported by: Matt Jordan ........ Merged revisions 400421 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	3422314cb2	Fix assumption in bridge_native_rtp.c regarding number of participants in a bridge. ... When a party leaves a bridge, there may be more participants in the bridge than expected. As such, it is important not to make assumptions regarding the list of channels in a bridge. This change makes it so that when a party leaves a native RTP bridge, we unbridge it and the party it was bridged with. Previously, the first and last channels in the list were unbridged since it was assumed that these were the two channels that had been bridged. As previously stated, a new party had been inserted into the bridge, so this logic did not work properly. (closes issue ASTERISK-22615) reported by Matt Jordan (closes issue ASTERISK-22532) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2899 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	3735f13434	Fix a crash caused by muting and unmuting a channel in ARI without specifying a direction. ... (closes issue ASTERISK-22637) Reported by: Scott Griepentrog Patch by Matt Jordan, whose office I have taken over in the name of Canada. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400401 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	0dd84b2f51	cel: Some whitespace cleanups ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400398 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	e59675b597	res_rtp_multicast: Ensure SSRC is set properly ... This fixes a bug where the SSRC field on multicast RTP can be stuck at 0 which can cause problems for endpoints trying to make sense of incoming streams. (closes issue ASTERISK-22567) Reported by: Simone Camporeale Patches: 22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536) ........ Merged revisions 400393 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400394 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	315ec8d7f9	Detect and use xsltCleanupGlobals when available ... This introduces usage of an additional libxslt cleanup function, xsltCleanupGlobals, when the configure script detects that it is available. Early versions of the library did not include this function. (closes issue ASTERISK-22570) Reported by: Corey Farrell Patches: xsltCleanupGlobals.patch uploaded by Corey Farrell (License 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	60423755e5	Update Alembic database scripts for external scripting and PostgreSQL, Oracle ... This patch does the following: 1) The env scripts have been updated to be tolerant of a NULL configuration file. This occurs when configuration is provided by an external script, such that the actual config.ini file is not used. 2) Enum types have all been given names. This is needed for PostgreSQL script generation. 3) The identifier meetme_confno_starttime_endtime is greater than 30 characters, and hence invalid for Oracle databases. This has been truncated down to meetme_confno_start_end. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	72fbce14f4	chan_vpb: Make compile again. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	3c2d7934b1	Get rid of uses of stasis_topic_wait() ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	024f9bf283	Fix crashes in res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and external_media_address is set. ... The callback function for changing the media address in streams wrongly assumes that a connection line will always be present. This is false as no line is present if a stream has been rejected. (closes issue ASTERISK-22645) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	23cea9e44b	Cache string values of formats on ast_format_cap() to save processing. ... Channel snapshots have string representations of the channel's native formats. Prior to this change, the format strings were re-created on ever channel snapshot creation. Since channel native formats rarely change, this was very wasteful. Now, string representations of formats may optionally be stored on the ast_format_cap for cases where string representations may be requested frequently. When formats are altered, the string cache is marked as invalid. When strings are requested, the cache validity is checked. If the cache is valid, then the cached strings are copied. If the cache is invalid, then the string cache is rebuilt and copied, and the cache is marked as being valid again. Review: https://reviewboard.asterisk.org/r/2879 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago