asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	fcb055fb4e	addition of G.719 pass-through support (closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	0bf94685fd	MSG_OOB flag on HANGUP packet removed. Per Tilghman's request on IRC (#asterisk-bugs). (closes issue #17506) Reported by: brycebaril Tested by: pabelanger, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	fb71a38a41	Don't send files twice and remove extra \r\n from header After the manager http auth changes, we forgot to remove the manual sending of the file. Also, ast_http_send adds two \r\n to the header that is passed to it, so a trailing \r\n is removed from the Content-type header. It might be better to change ast_http_send, but I don't like changing the behavior of an API function. (closes issue #17239) Reported by: cjacobsen Patches: patch2.diff uploaded by cjacobsen (license 1029) Tested by: lathama, cjacobsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a0d8a35659	Argh, mixed declarations and code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	81c15adfa2	Add distributed devicestate via the XMPP protocol. (closes issue #15757) Reported by: Marquis Patches: distributed_devstate-XMPP.txt uploaded by lmadsen (license 10) Tested by: Marquis, lmadsen, marcelloceschia Review: https://reviewboard.asterisk.org/r/351/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	41d7b51bf7	Merged revisions 270331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270331 \| pabelanger \| 2010-06-14 17:31:59 -0400 (Mon, 14 Jun 2010) \| 14 lines Properly play first file in sort list. When using sort=alpha we would always skip the first file in the list first time through. We now check for that properly. (closes issue #17470) Reported by: pabelanger Patches: sort.aplha.patch uploaded by pabelanger (license 224) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/703/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e98835e437	fixes crash in moh when cachertclasses flag is used The result for moh_register was not verified to guarantee the mohclass as added to the container. (closes issue #16993) Reported by: dmitri Patches: res_musiconhold_rtclass2.patch uploaded by dmitri (license 1001) moh_crash2.diff uploaded by dvossel (license 671) Tested by: dmitri git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	c672763af8	Fix some doxygen warnings. (closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	523e4e50bf	Release list lock before returning on error. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	857814f435	Add SRTP support for Asterisk 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	16 years ago
Tilghman Lesher	6279bf10c4	Avoid unloading res_smdi twice. (closes issue #17237) Reported by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	a55820a26d	Use the correct ical.h file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	464e44e325	Don't register functions until the last possible point, so they're not 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	16 years ago
Terry Wilson	9a2f04ce26	Fix ical library handling (again) 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	16 years ago
Mark Michelson	529a87ce7d	Remove unrelated MOH change from previous commit. Thanks Kevin! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	8999372c33	Fix misspelling of macro args. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	35025c16d0	Merged revisions 265910 via svnmerge from 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	16 years ago
Tilghman Lesher	5c9fdd8666	Construct socket name, according to the Postgres docs, and document as such. (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	16 years ago
Terry Wilson	f1503b9e1d	Ensure that libneon > 0.29.0 is installed for res_calendar_ews 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	16 years ago
Tilghman Lesher	a7498ae02e	Use configure to determine the prefixes and include directories properly. 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	16 years ago
Terry Wilson	880cde12ac	Calendaring support for Exchange Server 2007+ via EWS 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	16 years ago
Mark Michelson	0a63e3fa10	Log spandsp's fax debug output to the FAX logger level. Review: https://reviewboard.asterisk.org/r/658 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	51e7ee235b	fixes crash during dtmf 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	16 years ago
Tilghman Lesher	a21192f4a7	Make happy green color come back git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	113c677257	For FreeBSD git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	88a8703c37	Hmmm, probably should have read the manpage more thoroughly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8d6ee962c7	Add kqueue(2) implementation to Asterisk in various places. 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	16 years ago
Leif Madsen	c17cda109a	Revert previous WARNING message removal. 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	16 years ago
Leif Madsen	881450ec82	Remove unnecessary WARNING message in ael/pval.c (closes issue #17145) Reported by: okrief git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
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

1 2 3 4 5 ...

1497 Commits (531290385cd3ebdcb82e915135057d34c3fb480e)