1468 Commits (093dbfdd522bd0ca0c4b07c4f07e9ebdd6b8be21)

Author	SHA1	Message	Date
Jason Parker	d8dea9e76a	Merged revisions 262421 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | 11 lines Use a less silly method for modifying a flex-generated file. The sed syntax that was used wasn't actually valid, causing some versions to choke. This is the method that is used in 1.6.x+ for similar changes. (closes issue #16696) Reported by: bklang Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested by: qwell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262422 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a554e00fed	Merged revisions 260345 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it. The issue here is that in several cases, the channel was not locked while checking the stream. The result was that if two threads checked the state of the channel's stream at approximately the same time, then there could be a situation where both threads attempt to call ast_closestream on the channel's stream. The result here is that the refcount for the stream would go below 0, resulting in a crash. I have added proper channel locking to res_musiconhold.c to ensure that we do not try to check chan->stream without the channel locked. A Digium customer has been using this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	7f5a3370ad	Fix compile on systems without HAVE_NULLSAFE_PRINTF defined. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	13f523731a	Update res_fax and res_fax_spandsp to be compatible with Fax For Asterisk 1.2. ... The fax session initilization code for T.38 faxes has been rewritten. T.38 session initialization was removed from generic_fax_exec, and split into two different code paths for receive and send. Also the 'z' option (to send a T.38 reinvite if we do not receive one) was added to sendfax. In the output of 'fax show sessions', the 'Type' column has been renamed to 'Tech' and replaced with a new 'Tech' column that will report 'G.711' or 'T.38'. Control of ECM defaults has been added to res_fax A 'fax show settings' CLI command has been added. Support of the new AST_T38_REQUEST_PARMS control method request to handle channels that have already received a T.38 reinvite before the FAX application is start has been added. Support for the 'fax show settings' command has been added to res_fax_spandsp and handling of the ECM flag has been slightly altered. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	56a6994310	Merged revisions 258775 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) | 6 lines When StopMonitor is called, ensure that it will not be restarted by a channel event. (closes issue #16590) Reported by: kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	9e3f5fa6fb	Remove ABI differences that occured when compiling with DEBUG_THREADS. ... "Bad Things" would happen if Asterisk was compiled with DEBUG_THREADS, but a loaded module was not (or vice versa). This also immensely simplifies the lock code, since there are no longer 2 separate versions of them. Review: https://reviewboard.asterisk.org/r/508/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	f905bb1c0f	Fix the \brief description in the res_calendar_*.c files. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Julian Lyndon-Smith	d85650e4aa	Added MixMonitorMute manager command ... Added a new manager command to mute/unmute MixMonitor audio on a channel. Added a new feature to audiohooks so that you can mute either read / write (or both) types of frames - this allows for MixMonitor to mute either side of the conversation without affecting the conversation itself. (closes issue #16740) Reported by: jmls Review: https://reviewboard.asterisk.org/r/487/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	a5a0a5f867	Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number. ... SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	2be88e05c0	Allow symbol export filtering to work properly on platforms that have symbol prefixes. ... Some platforms prefix externally-visible symbols in object files generated from C sources (most commonly, '_' is the prefix). On these platforms, the existing symbol export filtering process ends up suppressing all the symbols that are supposed to be left visible. This patch allows the prefix string to be supplied to the top-level Makefile in the LINKER_SYMBOL_PREFIX variable, and then generates the linker scripts as required to include the prefix supplied. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bd716c50fd	Recorded merge of revisions 254452 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection. Here is a copy and paste of the details from my request on reviewboard that dealt with these changes: Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too: seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF. Fix 2. The second change in place is to fix an issue like the following: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end) *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list. Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254454 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	42577406fd	Improve handling of T.38 re-INVITEs that arrive before a T.38-capable ... application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame *can* fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	0eb71bccf1	handle_speechset has 4 arguments. ... Update code to reflect that handle_speechset has 4 arguments. (closes issue #17093) Reported by: gpatri Patches: res_agi.patch uploaded by gpatri (license 1014) Tested by: pabelanger, mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	5990fe07b8	Merged revisions 254235 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) | 72 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248860. As such the dialplan test has been extended: ; non absolute path, not combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test) exten => 5040, n, dial(sip/5001) ; absolute path, not combined exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten => 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1, monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ; combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m) exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n, dial(sip/5001) ; combined: changemonitor from no path to non absolute path exten => 5044, 1, monitor(wav,monitor_test6,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this wasn't possible before exten => 5044, n, dial(sip/5001) ; non absolute path, combined exten => 5045, 1, monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n, dial(sip/5001) ; absolute path, combined exten => 5046, 1, monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n, dial(sip/5001) ; no path, combined exten => 5047, 1, monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ; combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5048, 1, monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n, changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n, dial(sip/5001) ; combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5049, 1, monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n, changemonitor(tmp/jeff/monitor_test14) exten => 5049, n, dial(sip/5001) ; combined: changemonitor from no path to absolute exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n, changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n, dial(sip/5001) ; combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5051, 1, monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n, changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ; not combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19) exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n, dial(sip/5001) ; not combined: changemonitor from no path to non absolute exten => 5053, 1, monitor(wav,monitor_test21) exten => 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n, dial(sip/5001) ; not combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5054, 1, monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n, changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n, dial(sip/5001) ; not combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5055, 1, monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n, changemonitor(tmp/jeff/monitor_test25) exten => 5055, n, dial(sip/5001) ; not combined: changemonitor from no path to absolute exten => 5056, 1, monitor(wav,monitor_test26) exten => 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056, n, dial(sip/5001) ; not combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5057, 1, monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n, changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ae6008ef3a	Change per-file debug and verbose levels to be per-module, the way ... users expect them to work. 'core set debug' and 'core set verbose' can optionally change the level for a specific filename; however, this is actually for a specific source file name, not the module that source file is included in. With examples like chan_sip, chan_iax2, chan_misdn and others consisting of multiple source files, this will not lead to the behavior that users expect. If they want to set the debug level for chan_sip, they want it set for all of chan_sip, and not to have to also set it for reqresp_parser and other files that comprise the chan_sip module. This patch changes this functionality to be module-name based instead of file-name based. To make this work, some Makefile modifications were required to ensure that the AST_MODULE definition is present in each object file produced for each module as well. Review: https://reviewboard.asterisk.org/r/574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Philippe Sultan	5200b6e81e	Prevent a crash when a buddy gets offline. ... (closes issue #16760) Reported by: fiddur Patches: 248394.diff uploaded by fiddur (license 678)i with modifications by me Tested by: fiddur, phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	5f3730df4c	Include an extra newline after "Aliased CLI command" to get back the prompt. ... The other issue mentioned in this bug will be more difficult to resolve since we have no idea (right now) of knowing if the command that is aliased has been installed yet. (issue #16978) Reported by: jw-asterisk Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	43d922b5a6	Improve handling of values supplied to FAXOPT(ecm). ... Previously, values that began with whitespace were silently treated as 'no', and all non-'yes' values were also treated as 'no'. Now the supplied value is specifically checked for a 'yes' or 'no' (or equivalent) value, after skipping leading whitespace. If the value is not valid, then a warning message is generated. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	68d1ded8dd	Only change the RTP ssrc when we see that it has changed ... This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	7f29269d68	Merged revisions 250786 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250786 | jpeeler | 2010-03-04 19:02:58 -0600 (Thu, 04 Mar 2010) | 9 lines Fix not being able to specify a URL in MOH class directory. Don't attempt to chdir on a URL! (closes issue #16875) Reported by: raarts Patches: moh-http.patch uploaded by raarts (license 937) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	8ef8706944	Updated CHANGES file to mention res_fax and res_fax_spandsp. ... Also fixed MODULEINFO depends and conflicts for app_fax, res_fax, and res_fax_spandsp. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	06dc8bc123	Merge res_fax and res_fax_spandsp. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	06041ea28d	Fix several XML documentation validate errors. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249892 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	406bb18127	Merged revisions 248860 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) | 18 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248757. As such the dialplan test has been extended: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n, changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001) exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning exten => 5044, n, dial(sip/5001) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	d64987f8ad	Merged revisions 248757 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines Ensure that monitor recordings are written to the correct location. Recordings should be placed in the monitor directory when a non-absolute path is used. Exact dialplan used for testing: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) ABE-2101 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	e8df30b584	Improve support for RTCP reports without report blocks ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	de1d19f511	Revert an errant part of a previous cleanup, to fix a memory corruption issue. ... (closes issue #16368) Reported by: thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf (license 955) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Philippe Sultan	945529cae8	Add a new manager event for our buddies status. ... The new JabberStatus event gives a concise view of the status change to the AMI clients. Thanks fiddur! (closes issue #16760) Reported by: fiddur Patches: 244498.2.diff uploaded by fiddur (license 678) Tested by: fiddur, phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	83b0d30de5	res_pktccops needs to be able to export a symbol for chan_mgcp ... (closes issue #16782) Reported by: nahuelgreco Patches: res_pktccops.exports uploaded by nahuelgreco (license 162) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c8abb42e6a	Solaris doesn't like outputting a NULL to a %s in format strings. ... Detect all platforms that don't like that, either, and ensure that when documentation is missing, we pass a non-NULL pointer when outputting the corresponding documentation. (closes issue #16689) Reported by: bklang Patches: 20100209__issue16689__with_tests.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/497/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	6ad3619189	Fix crash on 32-bit for users not using https ... (closes issue #16778) Reported by: pitel Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, pitel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	1ffdf5c2ee	Merged revisions 242969 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242969 | tilghman | 2010-01-25 15:50:22 -0600 (Mon, 25 Jan 2010) | 2 lines Err, and use the new menuselect define, too. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242971 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	245bd1861f	Merged revisions 242852 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242852 | tilghman | 2010-01-25 14:15:45 -0600 (Mon, 25 Jan 2010) | 2 lines Restore FreeBSD to able-to-compile-ish-mode ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	fb65a6860a	Fix INTERNAL_OBJ error on stop when calendars.conf missing ... Initialize the calendars container before calling load_config and return FAILURE on allocation failure. Also, use the AST_MODULE_LOAD_* values for return values. Thanks to rmudgett for pointing out the error and the need to use the defined values for return git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	afb9fab574	Merged revisions 242728 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242728 | tilghman | 2010-01-24 23:42:22 -0600 (Sun, 24 Jan 2010) | 2 lines Buildbot pointed out an error (thanks, buildbot!) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	137046e459	Merged revisions 242723 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242723 | tilghman | 2010-01-24 23:33:37 -0600 (Sun, 24 Jan 2010) | 2 lines Oops, should have used CMD_PREFIX, not ECHO_PREFIX, for the commands. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	8f356343dd	Instead of crashing, allocate our header ast_str before we try to use it. ... (closes issue #16680) Reported by: lmadsen Patches: issue16680_20100122.patch uploaded by seanbright (license 71) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	bc9f02a60d	Merged revisions 242520 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script. Changed after discussion on the -dev list about possible unnecessary build failures, due to checkouts/untars causing these special source files to possibly be newer than their resulting C files. This should additionally ensure that nobody need learn about extra Makefile arguments to ensure the proper files get rebuilt when changes are made to these special source files. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3d51b9025f	Merged revisions 242423 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242423 | tilghman | 2010-01-22 15:44:18 -0600 (Fri, 22 Jan 2010) | 7 lines Rebuild from flex, bison sources when necessary. (issue #14629) Reported by: Marquis Patches: 20100121__issue14629.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e469483d82	rtp timestamp to timeval calculation fix ... The rtp timestamp to timeval calculation was only accurate for 8kHz audio. This patch corrects this. Review: https://reviewboard.asterisk.org/r/468/ SWP-648 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	49bf540c71	Create iterative method for querying SRV results, and use that for finding AGI servers. ... (closes issue #14775) Reported by: _brent_ Patches: 20091215__issue14775.diff.txt uploaded by tilghman (license 14) hagi-5.patch uploaded by brent (license 388) Tested by: _brent_ Reviewboard: https://reviewboard.asterisk.org/r/378/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	1a30c33d47	Get MoH building on OpenSolaris. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	a60935478b	Clarify error message in res_timing_timerfd. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	e612d87695	Convert a few places to use ast_calloc_with_stringfields where applicable. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	37b73283ff	Plug a memory leak in res_config_ldap. ... (closes issue #16257) Reported by: nito Patches: issue16257_20100111.diff uploaded by seanbright (license 71) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	cd2ba30e9e	If we aren't running on a machine that support CLOCK_MONOTONIC, don't load. ... Group developed and tested by seanbright, Corydon76, Kobaz, and Amorsen. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	a5688e483b	Fix crash in res_config_ldap. ... We need to allocate enough room for 2 pointers, not 2 characters. (closes issue #16397) Reported by: bklang Patches: res_config_ldap.patch uploaded by applsplatz (license 949) Tested by: applsplatz git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	cd8aa003e4	Socket level option is SOL_SOCKET, not SO_SOCKET. ... (issue #16580) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	ea47ab7ed9	Pass NULL for the ao2_callback function pointer instead of duplicating cb_true. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	055cb0e0ae	Add the class actually used in the MusicOnHold start event. ... (closes issue #16499) Reported by: syspert Patches: mohclass.patch uploaded by syspert (license 938) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago