asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	89bbecc724	Fix premature free'ing of the frame committed in r349608 Even though we set the frame to the ast_null_frame and return that, the caller of the frame hook may still need the frame. This now is a bit more careful about when it frees the frame, i.e., only under the same conditions that applied when we duplicated it in the first place. ........ Merged revisions 349822 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	12e3f412b5	Free successfully translated frame in fax_gateway_framehook A frame that is translated via ast_translate is also duplicated via ast_frdup. This will allocate a new frame on the heap, which needs to be free'd at the appropriate time. This issue reporter used valgrind to find that this occurred in res_fax's fax_gateway_framehook; a quick search through the code showed that only place this was currently not handling the translatted frame properly. (closes issue ASTERISK-19133) Reported by: Sylvain Rochet ........ Merged revisions 349608 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	fdda494776	Improve T.38 gateway V.21 preamble detection. This commit removes the V.21 preamble detection code previously added to the generic DSP implementation in Asterisk, and instead enhances the res_fax module to be able to utilize V.21 preamble detection functionality made available by FAX technology modules. This commit also adds such support to res_fax_spandsp, which uses the Spandsp modem tone detection code to do the V.21 preamble detection. There should be no functional change here, other than much more reliable V.21 preamble detection (and thus T.38 gateway initiation). ........ Merged revisions 349248 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	d9651f2be9	Fix timing source dependency issues with MOH Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patches: asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026) Review: https://reviewboard.asterisk.org/r/1578/ ........ Merged revisions 349194 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349195 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	78b17e6d41	Add a separate buffer for SRTCP packets The function ast_srtp_protect used a common buffer for both SRTP and SRTCP packets. Since this function can be called from multiple threads for the same SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the packets to become corrupted as the buffer was used by both threads simultaneously. This patch adds a separate buffer for SRTCP packets to avoid the problem. (closes issue ASTERISK-18889, Reported/patch by Daniel Collins) ........ Merged revisions 347995 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347996 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b05d4603c4	Fix crash during CDR update. The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ ........ Merged revisions 348362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348363 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	1c78d82f18	Don't clear LOCALSTATIONID before sending or receiving. The user may set that variable. ASTERISK-18921 ........ Merged revisions 348212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348213 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	ae61df53f1	Fix chan_jingle/gtalk load regression introduced in r346087 Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy for usage outside res_jabber. Testing of these changes focused on res_jabber itself, so this problem was missed. Reported-by: Michael Spiceland ........ Merged revisions 346951 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346952 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	83cd844b82	Re-resolve the STUN address if a STUN poll fails for res_stun_monitor. The STUN socket must remain open between polls or the external address seen by the STUN server is likely to change. However, if the STUN request poll fails then the STUN server address needs to be re-resolved and the STUN socket needs to be closed and reopened. * Re-resolve the STUN server address and create a new socket if the STUN request poll fails. * Fix ast_stun_request() return value consistency. * Fix ast_stun_request() to check the received packet for expected message type and transaction ID. * Fix ast_stun_request() to read packets until timeout or an associated response packet is found. The stun_purge_socket() hack is no longer required. * Reduce ast_stun_request() error messages to debug output. * No longer pass in the destination address to ast_stun_request() if the socket is already bound or connected to the destination. (closes issue ASTERISK-18327) Reported by: Wolfram Joost Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1595/ ........ Merged revisions 346700 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346701 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	77b670c4ab	Allow each logging destination and console to have its own notion of the verbosity level. Review: https://reviewboard.asterisk.org/r/1599 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	edaf970c38	Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified. (closes issue ASTERISK-18693) Reported by: Davide Dal Fra Review: https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter Doekes ........ Merged revisions 346292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346293 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	e6ca768081	Fix res_jabber resource leaks This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346087 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	6d05a31d9f	Resume playing existing hold music for cached realtime MOH As a result of the fix for ASTERISK-18039, realtime caching MOH no longer properly resumes playing back a file between different holds in the same call. This is because scanning for new files causes the existing file array to be emptied and we were just comparing that the saved pointer to the filename matched the pointer to the filename in a particular position in the array. An easy fix is to save the filename instead of a pointer to it and then do a strcmp instead of comparing the addresses. (closes issue ASTERISK-18912) Review: https://reviewboard.asterisk.org/r/1596/ ........ Merged revisions 346030 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346031 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	f59322f724	Added support level for new modules ........ Merged revisions 346029 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a86037d959	Make FastAGI HANGUP show up in AGI debug output. * Change from using send() to ast_agi_send() so the HANGUP shows up in the AGI debug output. (closes issue ASTERISK-18723) Reported by: James Van Vleet Patches: jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 345431 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345432 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	bd486fcf41	Don't forget to rescan MOH files for cached realtime classes Realtime MOH class caching was implemented because without it, you would build a completely new MOH class and would start the music over at the beginning each time hold was pressed in a conversation. Unfortunately, this broke re-scanning for file changes for realtime MOH classes. This patch corrects that issue. (closes issue ASTERISK-18039) Review: https://reviewboard.asterisk.org/r/1579/ ........ Merged revisions 344899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344900 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	3d44965e70	only attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses Patch by: jkonieczny (modified) ASTERISK-18490 ........ Merged revisions 344330 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344334 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	969f4aa3d6	Fix sqlite config driver segfault and broken queries The sqlite realtime handler assumed you had a static config configured as well. The realtime multientry handler assumed that you weren't using dynamic realtime. (closes issue ASTERISK-18354) (closes issue ASTERISK-18355) Review: https://reviewboard.asterisk.org/r/1561 ........ Merged revisions 343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343393 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343394 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	25ee5f83b5	Cleanup references to sipusers and sipfriends dynamic realtime families Somewhere between 1.4 and 1.8 the sipusers family has become completely unused. Before that, the sipfriends family had been obsoleted in favor of separate sipusers and sippeers families. Apparently, they have been merged back again into a single family which is now called "sippeers". Reviewed by: irroot, oej, pabelanger Review: https://reviewboard.asterisk.org/r/1523 ........ Merged revisions 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342870 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342871 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	4b826c46b3	Don't crash on empty notify channel ........ Merged revisions 342715 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	e5ac65bb43	Fix sequence number overflow over 16 bits causing codec change in RTP packets. Sequence number was handled as an unsigned integer (usually 32 bits I think, more depending on the architecture) and was put into the rtp packet which is basically just a bunch of bits using an or operation. Sequence number only has 16 bits allocated to it in an RTP packet anyway, so it would add to the next field which just happened to be the codec. This makes sure the sequence number is set to be a 16 bit integer regardless of architecture (hopefully) and also makes it so the incrementing of the sequence number does bitwise or at the peak of a 16 bit number so that the value will be set back to 0 when going beyond 65535 anyway. (closes issue ASTERISK-18291) Reported by: Will Schick Review: https://reviewboard.asterisk.org/r/1542/ ........ Merged revisions 342602 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342603 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	b61256c64b	Cleanup reference leaks in res_jabber res_jabber.c had a number of places where astobjs would be referenced and have their reference counts bumped without having a dereference made before the object lost scope. This patch adds a number of ASTOBJ_UNREFs to resolve that. Review: https://reviewboard.asterisk.org/r/1478/ ........ Merged revisions 342545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342546 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	b009ea5216	White space fixes in res_fax git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b961d57c4c	Fix AGI exec Park to honor the Park application parameters. The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341718 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	4b9546abdf	Merged revisions 340971 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340971 \| kmoore \| 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) \| 15 lines Merged revisions 340970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 \| kmoore \| 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) \| 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	9d83162d55	Don't skip the query field on a realtime multi query There is no documented reason to not add the query field to the varlist returned by a realtime multi query, despite the config category being set to its value. Of course, there is no documentation that the category should be set to the value either. There is lots of no documentation when it comes to realtime. But, other engines do not skip this field so I am forcing this backend to follow the convention, because not doing so is very silly. ........ Merged revisions 340662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340663 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	bb07ca66a1	Merged revisions 340109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340109 \| mnicholson \| 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) \| 18 lines Merged revisions 340108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 \| mnicholson \| 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) \| 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	07133b3a96	Merged revisions 339507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339507 \| mnicholson \| 2011-10-05 11:32:59 -0500 (Wed, 05 Oct 2011) \| 10 lines Merged revisions 339505 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339505 \| mnicholson \| 2011-10-05 11:31:21 -0500 (Wed, 05 Oct 2011) \| 3 lines The app name in the documentation must match what we register the application as. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	b698038995	Add generic faxdetect framehook to res_fax Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no to enable dialplan faxdetect allowing more flexibility. as soon as a fax tone is detected the framehook is removed. there is a penalty involved in running this framehook on non G711 channels as they will be transcoded. CNG tone is suppresed using the SQUELCH flag to allow WaitForNoise to be run on the channel to detect Voice. (Closes issue ASTERISK-18569) Reported by: Myself Reviewed by: Matthew Nicholson, Kevin Fleming Review: https://reviewboard.asterisk.org/r/1116/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	1b3bd7ddb4	Merged revisions 339463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339463 \| irroot \| 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) \| 9 lines Only change the capabilities on the gateway when the session is been destroyed there is still a race condition that ends in a segfault. if the caps are changed the logic in res_fax_spandsp will run T30 code not gateway code to end the session. this has been experienced on a "slower" under spec system. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	635118043d	Merged revisions 339298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339298 \| jrose \| 2011-10-04 09:09:50 -0500 (Tue, 04 Oct 2011) \| 19 lines Merged revisions 339297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339297 \| jrose \| 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) \| 13 lines Reverting revision 333265 due to component connection problems it introduces. I'm going to attempt some generic res_jabber cleanup and come up with a new fix for this problem, but first it seems prudent to remove this rather broad attempt to fix it and instead approach this problem either from the same angle but looking only at canceling (or possibly rescheduling) the send when we absolutely know it will cause a segfault or, if that can't be easily accomplished, strictly from the devstate side of things. Also, I'm pretty sure a lot of the code in res_jabber isn't thread safe. (issue ASTERISK-18626) (issue ASTERISK-18078) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	69ea68a1f5	Merged revisions 339045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339045 \| mnicholson \| 2011-10-03 10:54:55 -0500 (Mon, 03 Oct 2011) \| 4 lines Ported ast_fax_caps_to_str() to 10, not sure why it wasn't already here. This function prints a list of caps instead of a hex bitfield. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	0932d899e6	Merged revisions 339043 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339043 \| mnicholson \| 2011-10-03 10:41:36 -0500 (Mon, 03 Oct 2011) \| 2 lines Don't clear the AST_FAX_TECH_MULTI_DOC flag right after we set it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	9a5de09f92	Merged revisions 339011 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339011 \| mnicholson \| 2011-10-03 10:19:44 -0500 (Mon, 03 Oct 2011) \| 2 lines properly remove the AST_FAX_TECH_GATEWAY flag (instead of setting all of the other flags) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	ebf3632e08	Merged revisions 338950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338950 \| irroot \| 2011-10-03 11:37:59 +0200 (Mon, 03 Oct 2011) \| 14 lines Fixup a race condition in res_fax.c where FAXOPT(gateway)=no will turn off the gateway but the framehook is not destroyed. this problem happens when a gateway is attempted in the dialplan and the device is not available i may want to do fax to mail in the server it will not be allowed. instead of checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts 338904 Fix some white space. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	b5147c8817	Merged revisions 338904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338904 \| irroot \| 2011-10-02 16:17:32 +0200 (Sun, 02 Oct 2011) \| 8 lines Remove T38 Gateway capability when detaching framehook. SET(FAXOPT(gateway)=no) does not remove the capability when detaching the framehook. small patch to fix this problem. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	c04ab6b35c	Just formatting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8a74aa9ef9	Merged revisions 337542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337542 \| irroot \| 2011-09-22 13:44:22 +0200 (Thu, 22 Sep 2011) \| 14 lines Merged revisions 337541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337541 \| irroot \| 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) \| 8 lines Add warned to ast_srtp to prevent errors on each frame from libsrtp The first 9 frames are not reported as some devices dont use srtp from first frame these are suppresed. the warning is then output only once every 100 frames. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	2ae7ae00c8	Merged revisions 337178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337178 \| oej \| 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) \| 14 lines Change strictrtp option to default to yes in the RTP module Suggested by Kapejod on Facebook Review: https://reviewboard.asterisk.org/r/1448/ (closes issue ASTERISK-18587) Thanks for quick feedback to kpfleming and Tilghman --Denna och nedanstående rader kommer inte med i loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M res/res_rtp_asterisk.c ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	14d3f891e0	Merged revisions 336878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336878 \| russell \| 2011-09-19 20:03:55 -0500 (Mon, 19 Sep 2011) \| 43 lines Merged revisions 336877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336877 \| russell \| 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) \| 36 lines Fix crashes in ast_rtcp_write(). This patch addresses crashes related to RTCP handling. The backtraces just show a crash in ast_rtcp_write() where it appears that the RTP instance is no longer valid. There is a race condition with scheduled RTCP transmissions and the destruction of the RTP instance. This patch utilizes the fact that ast_rtp_instance is a reference counted object and ensures that it will not get destroyed while a reference is still around due to scheduled RTCP transmissions. RTCP transmissions are scheduled and executed from the chan_sip scheduler context. This scheduler context is processed in the SIP monitor thread. The destruction of an RTP instance occurs when the associated sip_pvt gets destroyed (which happens when the sip_pvt reference count reaches 0). However, the SIP monitor thread is not the only thread that can cause a sip_pvt to get destroyed. The sip_hangup function, executed from a channel thread, also decrements the reference count on a sip_pvt and could cause it to get destroyed. While this is being changed anyway, the patch also removes calling ast_sched_del() from within the RTCP scheduler callback. It's not helpful. Simply returning 0 prevents the callback from being rescheduled. (closes issue ASTERISK-18570) Related issues that look like they are the same problem: (issue ASTERISK-17560) (issue ASTERISK-15406) (issue ASTERISK-15257) (issue ASTERISK-13334) (issue ASTERISK-9977) (issue ASTERISK-9716) Review: https://reviewboard.asterisk.org/r/1444/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	364eb56835	Merged revisions 336717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336717 \| jrose \| 2011-09-19 15:16:23 -0500 (Mon, 19 Sep 2011) \| 14 lines Merged revisions 336716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336716 \| jrose \| 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) \| 7 lines Document applications that play audio and do not answer unanswered calls. This patch is part of an effort to document early media and its usage. If you are interested in contributing to this documentation effort, there are probably other applications worth documenting as well as an Asterisk wiki article at https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	2a25779d47	Merged revisions 335510 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335510 \| russell \| 2011-09-13 02:24:34 -0500 (Tue, 13 Sep 2011) \| 22 lines Merged revisions 335497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335497 \| russell \| 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) \| 15 lines Fix a crash in res_ais. This patch resolves a crash observed in a load testing environment that involved the use of the res_ais module. I observed some crashes where the event delivery callback would get called, but the length parameter incidcating how much data there was to read was 0. The code assumed (with good reason I would think) that if this callback got called, there was an event available to read. However, if the rare case that there's nothing there, catch it and return instead of blowing up. More specifically, the change always ensure that the size of the received event in the cluster is always big enough to be a real ast_event. Review: https://reviewboard.asterisk.org/r/1423/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	1fed068bae	Add SQLite 3 realtime support git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	35e27201c7	Merged revisions 334357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334357 \| rmudgett \| 2011-09-02 16:08:16 -0500 (Fri, 02 Sep 2011) \| 26 lines Merged revisions 334355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334355 \| rmudgett \| 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) \| 19 lines MusicOnHold has extra unref which may lead to memory corruption and crash. The problem happens when a call is disconnected and you had started a MOH class that does not use the files mode. If you define REF_DEBUG and recreate the problem, it will announce itself with the following warning: Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained, and class is still in a container! * Fixed moh_alloc() and moh_release() functions not handling the state->class reference consistently. (closes issue ASTERISK-18346) Reported by: Mark Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski Review: https://reviewboard.asterisk.org/r/1404/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	e68be70646	Merged revisions 334230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334230 \| tilghman \| 2011-09-01 12:30:19 -0500 (Thu, 01 Sep 2011) \| 25 lines Merged revisions 334229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334229 \| tilghman \| 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011) \| 18 lines Create a local alias for ast_odbc_clear_cache. As a function pointer, the reference has to be resolved at load time irrespective of the RTLD_LAZY flag. Creating a local alias solves this problem, because the structure is initialized with that local function pointer, while the actual function can remain lazily linked until runtime. The reason why this is important is because we lazily load function references during the module loading process, in order to obtain priority values for each module, ensuring that modules are loaded in the correct order. Previous to this change, when this module was initially loaded, the module loader would emit a symbol resolution error, because of the above requirement. Closes ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by Walter Doekes, patch by me) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	dadc749dac	Merged revisions 334064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r334064 \| mnicholson \| 2011-08-31 11:31:00 -0500 (Wed, 31 Aug 2011) \| 4 lines only alter the gateway_timeout when attching the gateway to a channel ASTERISK-18219 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	cae7253575	Merged revisions 333895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r333895 \| mnicholson \| 2011-08-30 09:01:31 -0500 (Tue, 30 Aug 2011) \| 6 lines Replaced FAXOPT(gwtimeout) with a second parameter to FAXOPT(gateway). Patch by: irroot Review: https://reviewboard.asterisk.org/r/1385/ ASTERISK-18219 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	7067bb8b42	Merged revisions 333716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r333716 \| mnicholson \| 2011-08-29 13:22:58 -0500 (Mon, 29 Aug 2011) \| 5 lines It is possible for the gateway to be attached when the channel is still negotiating T.38. This change handles that case. ASTERISK-18329 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	d836c88b49	Merged revisions 333570 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333570 \| jrose \| 2011-08-29 10:56:56 -0500 (Mon, 29 Aug 2011) \| 11 lines Merged revisions 333569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333569 \| jrose \| 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) \| 4 lines Accidental use of variable client->status instead of client->state in from ASTERISK-18078 (issue ASTERISK-18078) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	10183c021e	Merged revisions 333410 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333410 \| jrose \| 2011-08-26 11:28:03 -0500 (Fri, 26 Aug 2011) \| 19 lines Merged revisions 333378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333378 \| jrose \| 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) \| 13 lines [patch] Buddies are always auto-registered when processing the roster Reporter said autoregister flag was ignored for registering 'buddies' which had a subscription to us. Verified that this was the case and observed how the patch addressed this and made sure it didn't break anything. (closes issue ASTERISK-14233) Reported by: Simon Arlott Patches: asterisk-0015229.patch (license #5756) patch uploaded by Simon Arlott Tested by: Jonathan Rose ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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