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2010-06-08Add SRTP support for AsteriskTerry Wilson
After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Avoid unloading res_smdi twice.Tilghman Lesher
(closes issue #17237) Reported by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-01Use the correct ical.h fileTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-01Don't register functions until the last possible point, so they're not ↵Tilghman Lesher
unloaded unnecessarily. (closes issue #15996) Reported by: junky Patches: sdmi_wait.diff uploaded by junky (license 177) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-28Fix ical library handling (again)Terry Wilson
Newer versions of libical (which we require) store the header file in a libical/ subfolder and include an ical.h file that does a #warning for deprecation and then #includes <libical/ical.h>. Since we now test for libical/ical.h, we can change the #includes back to <libical/ical.h> and remove the test which specifically adds /usr/include/libical as an include directory. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-26Remove unrelated MOH change from previous commit.Mark Michelson
Thanks Kevin! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-26Fix misspelling of macro args.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-26Merged revisions 265910 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010) | 7 lines Not finding rows in the DB does not rise to the level of a warning. (closes issue #17062) Reported by: drookie Patches: 20100525__issue17062.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-26Construct socket name, according to the Postgres docs, and document as such.Tilghman Lesher
(closes issue #17392) Reported by: dps Patches: 20100525__issue17392.diff.txt uploaded by tilghman (license 14) Tested by: dps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-26Ensure that libneon > 0.29.0 is installed for res_calendar_ewsTerry Wilson
This uses a modified version of pabelanger's patch that checks for NTLM support instead, which was added in 0.29.0 which is what is required for res_calendar_ews. (closes issue #17391) Reported by: loloski Patches: issue17391.patch.v2 uploaded by pabelanger (license 224) Tested by: twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-26Use configure to determine the prefixes and include directories properly.Tilghman Lesher
This ensures cross-platform compatibility, even among Linux distributions, which don't always put headers in the same place. (closes issue #17391) Reported by: loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-24Calendaring support for Exchange Server 2007+ via EWSTerry Wilson
This commit adds support for calendaring with Exchange Server 2007+ via Exchange Web Services. Full write support and for querying attendees. Many thanks to Jan Kaláb for the feature. (closes issue #17022) Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel (license 1008) Tested by: pitel, twilson Review: https://reviewboard.asterisk.org/r/557/ Review: https://reviewboard.asterisk.org/r/668/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-21Log spandsp's fax debug output to the FAX logger level.Mark Michelson
Review: https://reviewboard.asterisk.org/r/658 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19fixes crash during dtmfDavid Vossel
During the processing of Cisco dtmf the dtmf samples were not being calculated correctly. In an attempt to determine what sample rate was being used, a NULL frame was processed which caused a crash. This patch resolves this. (closes issue #17248) Reported by: falves11 Patches: issue_17248.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-18Make happy green color come backTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-13For FreeBSDTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-13Hmmm, probably should have read the manpage more thoroughly.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-13Add kqueue(2) implementation to Asterisk in various places.Tilghman Lesher
This will save a considerable amount of CPU on the BSDs, including Mac OS X, as it eliminates several places in the code that we previously used a busy loop. Additionally, this adds a res_timing interface, using kqueue timers. Review: https://reviewboard.asterisk.org/r/543/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-12Revert previous WARNING message removal.Leif Madsen
Marquis42 suggested a better method of doing what I wanted because I ended up removing the WARNING message for all instances when really I just wanted to remove it for the 'return' keyword, not everything. (issue #17145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-12Remove unnecessary WARNING message in ael/pval.cLeif Madsen
(closes issue #17145) Reported by: okrief git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-11Merged revisions 262421 via svnmerge from Jason Parker
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
2010-04-30Merged revisions 260345 via svnmerge from Mark Michelson
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
2010-04-27Fix compile on systems without HAVE_NULLSAFE_PRINTF defined.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-26Update res_fax and res_fax_spandsp to be compatible with Fax For Asterisk 1.2.Matthew Nicholson
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
2010-04-25Merged revisions 258775 via svnmerge from Tilghman Lesher
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
2010-04-22Remove ABI differences that occured when compiling with DEBUG_THREADS.Jason Parker
"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
2010-04-21Fix the \brief description in the res_calendar_*.c files.Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21Added MixMonitorMute manager commandJulian Lyndon-Smith
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
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.Richard Mudgett
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
2010-04-02Allow symbol export filtering to work properly on platforms that have symbol ↵Kevin P. Fleming
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
2010-03-25Recorded merge of revisions 254452 via svnmerge from Mark Michelson
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
2010-03-25Improve handling of T.38 re-INVITEs that arrive before a T.38-capableKevin P. Fleming
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
2010-03-25handle_speechset has 4 arguments.Leif Madsen
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
2010-03-24Merged revisions 254235 via svnmerge from Jeff Peeler
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
2010-03-23Change per-file debug and verbose levels to be per-module, the wayKevin P. Fleming
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
2010-03-18Prevent a crash when a buddy gets offline.Philippe Sultan
(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
2010-03-16Include an extra newline after "Aliased CLI command" to get back the prompt.Sean Bright
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
2010-03-15Improve handling of values supplied to FAXOPT(ecm).Kevin P. Fleming
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
2010-03-12Only change the RTP ssrc when we see that it has changedTerry Wilson
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
2010-03-05Merged revisions 250786 via svnmerge from Jeff Peeler
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
2010-03-03Updated CHANGES file to mention res_fax and res_fax_spandsp.Matthew Nicholson
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
2010-03-02Merge res_fax and res_fax_spandsp.Matthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02Fix several XML documentation validate errors.Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-25Merged revisions 248860 via svnmerge from Jeff Peeler
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
2010-02-25Merged revisions 248757 via svnmerge from Jeff Peeler
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
2010-02-20Improve support for RTCP reports without report blocksOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-18Revert an errant part of a previous cleanup, to fix a memory corruption issue.Tilghman Lesher
(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
2010-02-18Add a new manager event for our buddies status.Philippe Sultan
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
2010-02-10res_pktccops needs to be able to export a symbol for chan_mgcpTilghman Lesher
(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
2010-02-10Solaris doesn't like outputting a NULL to a %s in format strings.Tilghman Lesher
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
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-08 16:35:41 +0000