asterisk

Commit Graph

Author	SHA1	Message	Date
Terry Wilson	d81a8e34dd	Don't add Attendees during copy, replace them git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	a75ba8d1a9	Remove global variable that makes dlopen unhappy This isn't the best way to do this, but it is the easiest. There are some limitations that are going to need to be addressed at some point with reloads and when I (or someone else) work on that, then the API can be updated to handle passing the private config data that the calendar tech modules need in a better way as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223016 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	864aa14426	Formatting, moving error messages to ERROR, removing references to unexisting debug output. No functionality changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	fff998bf41	Use extref for doxygen references to external libraries (in this case PostgreSQL) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222614 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2d60b75594	Change schema query to involve the use of an optional schema parameter. This change is done in such a way as to allow the driver to continue to function with older databases which don't have these features. (closes issue #16000) Reported by: jamicque Patches: 20091002__issue16000.diff.txt uploaded by tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	78012e4f71	When we call a gosub routine, the variables should be scoped to avoid contaminating the caller. This affected the ~~EXTEN~~ hack, where a subroutine might have changed the value before it was used in the caller. Patch by myself, tested by ebroad on #asterisk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	1c9fe00920	Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 \| kpfleming \| 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) \| 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	717d2ec3c9	Remove spurious debug git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	10ce6cd757	Use rtp properties instead of adding a callback Thanks, Josh. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	865daf4858	Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 \| twilson \| 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) \| 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Philippe Sultan	b11b94a083	Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channels JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over XMPP to process calls. SendText can be used instead of JabberSend in the context of XMPP based voice channels (chan_gtalk and chan_jingle). (closes issue #12569) Reported by: eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo Review: https://reviewboard.asterisk.org/r/88/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	d8a2d3dedf	Remove some unused defines from res_jabber. (closes issue #15359) Reported by: snuffy Patches: bug_res_jabber_unused_defines.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	3c04a79abf	use the actual given ip address for 'rtp set debug ip <foo>' instead of the word 'ip' (closes issue #15711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7) Tested by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218107 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthias Nick	8e1bae06bf	Sets the correct musicclass after an announcement	16 years ago
Tilghman Lesher	c9dd40c1f6	Verify support for wide ODBC character types before using them. (closes issue #15870) Reported by: nic_bellamy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	1ed1eb277e	gcc 4.4 fix: union instead of cast gcc 4.4 has more strict rules for aliasing. It doesn't like a struct sockaddr_in pointer pointing to a struct sockaddr. So we make it a union. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	fe7ec8c675	Remove what appears to be an unnecessary define. (closes issue #15851) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	eca8f9082c	Adding MUTEAUDIO() dialplan function and MuteAudio AMI action (pinepeach) Review: https://reviewboard.asterisk.org/r/345/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	fdd078af52	Remove unnecessary define for Solaris (closes issue #15358) Reported by: snuffy Patches: bug_res_moh_remove_unneeded_include.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	6f63f3eb8d	cast time_t type variables to long where needed. This makes res_calendar.c compile on OpenBSD and the same cast is used in a lot of other places where time_t type vars are used. (closes issue #15656) Reported by: mvanbaak Patches: 2009081100-rescalendarcompilefix.diff.txt uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Gavin Henry	f2b9fc797d	Added three new attributes and applied a patch to res_config_ldap.c attributetype ( AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent' DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) and patch fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725) Reported by: macogeek Patches: fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license 863) Tested by: suretec git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	642bec4d6f	AST-2009-005 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ed8ccbdb73	Gracefully handle malformed RTP text packets. AST-2009-004 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	33a48e257e	Honor channel's music class when using realtime music on hold. (closes issue #15051) Reported by: alexh Patches: 15051.patch uploaded by mmichelson (license 60) Tested by: alexh git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	d81d6d3415	Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	4ff3f0058d	Clarify documentation on 'realtime update2' to show more than one condition. (closes issue #15357) Reported by: snuffy Patches: bug_fix_doc_update2.diff uploaded by snuffy (license 35) (slightly modified by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	96e4e31eeb	Merged revisions 207647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207647 \| kpfleming \| 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) \| 12 lines Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker after all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided either in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4cf8a968fd	Add an API for reporting security events, and a security event logging module. This commit introduces the security events API. This API is to be used by Asterisk components to report events that have security implications. A simple example is when a connection is made but fails authentication. These events can be used by external tools manipulate firewall rules or something similar after detecting unusual activity based on security events. Inside of Asterisk, the events go through the ast_event API. This means that they have a binary encoding, and it is easy to write code to subscribe to these events and do something with them. One module is provided that is a subscriber to these events - res_security_log. This module turns security events into a parseable text format and sends them to the "security" logger level. Using logger.conf, these log entries may be sent to a file, or to syslog. One service, AMI, has been fully updated for reporting security events. AMI was chosen as it was a fairly straight forward service to convert. The next target will be chan_sip. That will be more complicated and will be done as its own project as the next phase of security events work. For more information on the security events framework, see the documentation generated from doc/tex/. "make asterisk.pdf" Review: https://reviewboard.asterisk.org/r/273/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ba2a8457b8	Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 \| dvossel \| 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) \| 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0e8c630224	Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	ae87ba45b5	Add support for multicast RTP paging. (closes issue #11797) Reported by: macbrody Review: https://reviewboard.asterisk.org/r/270/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6b53ec413d	Fix 2 typos and add support for wide character types. Reported by Benny Amorsen via the asterisk-users mailing list. http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	dcfe69ec64	fixes some memory leaks and redundant conditions (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	730e60e583	Merged revisions 201600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201600 \| russell \| 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) \| 29 lines Fix memory corruption and leakage related reloads of non files mode MoH classes. For Music on Hold classes that are not files mode, meaning that we are executing an application that will feed us audio data, we use a thread to monitor the external application and read audio from it. This thread also makes use of the MoH class object. In the MoH class destructor, we used pthread_cancel() to ask the thread to exit. Unfortunately, the code did not wait to ensure that the thread actually went away. What needed to be done is a pthread_join() to ensure that the thread fully cleans up before we proceed. By adding this one line, we resolve two significant problems: 1) Since the thread was never joined, it never fully goes away. So, on every reload of non-files mode MoH, an unused thread was sticking around. 2) There was a race condition here where the application monitoring thread could still try to access the MoH class, even though the thread executing the MoH reload has already destroyed it. (issue #15109) Reported by: jvandal (issue #15123) Reported by: axisinternet (issue #15195) Reported by: amorsen (issue AST-208) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	dce6a54a4a	Trunk implementation of setting an alternate RTP source. This contains the interface by which we can let an rtp instance know that it might start receiving audio from a new source. This is similar in nature to revision 197588 of Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	a179e144cd	Show the interface name on error, if it is not found. If the smdiport specified is not found, show the interface name instead of '(null)'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	82fb56886e	More 'static' qualifiers on module global variables. The 'pglobal' tool is quite handy indeed :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	6c5987811c	Redesigned 'optional API' support. This patch provides a new implementation of the optional API support defined in asterisk/optional_api.h; this new version provides solves compatibility issues with the use of linker version scripts for suppressing global symbols. In addition, there is now a functional (and tested!) implementation for Mac OS/X, so module writers no longer need to use special tests before calling optional API functions. All future implementations must provide these same semantics, so that module writers can rely on them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d532cbcd8a	module load priority This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	515166ba37	Move music on hold related applications documentation to XML. Move MusicOnHold, SetMusicOnHold, StartMusicOnHold, StopMusicOnHold static documentation to the new AstXML form. (issue #15245) Reported by: eliel Patches: res_musiconhold_static_conversion.txt uploaded by lmadsen (license 10) (with some fixes and formatting by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	4f94236de5	Move function PP_EACH_USER and PP_EACH_EXTENSION documentation to XML. Move function PP_EACH_USER and PP_EACH_EXTENSION documentation to the new AstXML form. (issue #15245) Reported by: eliel Patches: res_phoneprov_static_conversion.txt uploaded by lmadsen (license 10) (with PP_EACH_USER add by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	9ce385bd72	Move static docs to the new AstXML form. Move SMDI_MSG and SMDI_MSG_RETRIEVE functions statis documentation to XML. (issue #15245) Reported by: eliel Patches: res_smdi_static_conversion.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	fb73ee6187	Moved more static documentation to the new AstXML form. Moved more static docs to XML (pplications and manager actions): Monitor, StopMonitor, ChangeMonitor, PauseMonitor, UnpauseMonitor. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	8d464b7211	Move JabberSend manager action from static docs to the AstXML form. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	1b59a1cd7d	Move static documentation of E\|Dead\|AGI() application and manager action to XML. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	0c99bc31cb	Avoid a crash when res_timing_dahdi is unloaded but wasn't properly loaded. if dahdi_test_timer() fails, timing_funcs_handle remains NULL causing a crash when calling ast_unregister_timing_interface() with a NULL pointer. (closes issue #15234) Reported by: eliel Patches: timing_dahdi1.diff uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	3353710e16	Properly terminate the receive buffer before sending to iksemel. aji_io_recv takes the maximum number of bytes to read (instead of the total buffer size), so we have to subtract 1 from our buffer size. Without this, when we receive packets that are larger than our buffer, iksemel will choke and things get wonky. (closes issue #15232) Reported by: lp0 Patches: 05302009_res_jabber.c.patch uploaded by seanbright (license 71) Tested by: seanbright, lp0 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	90c3db40ed	Merged revisions 198370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198370 \| seanbright \| 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) \| 12 lines Properly terminate AMI JabberSend response messages. The response message (either Error or Success) needs an extra trailing \r\n after the fields to inform the client that the message is complete. (closes issue #14876) Reported by: srt Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71) asterisk_14876.patch uploaded by srt (license 378) trunk-14876-2.diff uploaded by phsultan (license 73) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1ee78437e4	Merged revisions 198311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198311 \| russell \| 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) \| 5 lines Fix a crash that occurred when MWI SMDI messages expired. (closes issue #14561) Reported by: cmoss28 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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1383 Commits (66e993de951dcb3fe536f51b6f4b77a2c5e52180)