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${ noResults }
21644 Commits (10f33d5e70d9bd59be05e80d9e078cd5a8fd26b7)
Author | SHA1 | Message | Date |
---|---|---|---|
|
10f33d5e70 |
Merged revisions 335320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6 addresses via DNS queries. (closes issue ASTERISK-18090) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335321 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
0f50837b20 |
Merged revisions 335259 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) | 6 lines build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value. adding an ao2_unlink from the peers_by_ip container fix it. Review: https://reviewboard.asterisk.org/r/1428/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335260 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
4e57652651 |
Merged revisions 335064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335078 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
d5544999b5 |
Move code for VALID_EXTEN from app_readexten to func_dialplan
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14 years ago |
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ce8cb2769e |
Merged revisions 334953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011) | 10 lines Fix crash with res_fax when MALLOC_DEBUG and "core stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is enabled when res_fax tries to unregister its logger level. * Make ast_logger_unregister_level() use ast_free() instead of free(). When MALLOC_DEBUG is enabled, ast_free() does not degenerate into a call to free(). Therefore, if you allocated memory with a form of ast_malloc you must free it with ast_free. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d4198b3ecf |
Merged revisions 334843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep 2011) | 4 lines Cleanup chan_iax2.c log messages Review: https://code.asterisk.org/code/cru/CR-AST-11 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 334840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines Fix AMI action Park crash. * Made AMI action Park not say anything to the parker channel (AMI header Channel2) since the AMI action is a third party parking the call. (This is a change in behavior that cannot be preserved without a lot of effort.) * Made not play pbx-parkingfailed if the Park 's' option is used. JIRA AST-660 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3badf91d58 |
Merged revisions 334682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334747 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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8211ef0f13 |
another clean up
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334746 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1e13555525 |
Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d1af4d4f15 |
Merged revisions 334620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) | 2 lines peroid typo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b8a0261ffb |
Merged revisions 334616 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep 2011) | 10 lines Prevent segfault if call arrives before Asterisk is fully booted. Prevent ast_pbx_start and ast_run_start from starting a new thread unless asterisk is fully booted. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1407/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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67d2ce9e73 |
authdebug is now disabled by default
To enable this functionaility again set authdebug = yes in iax.conf Review: https://reviewboard.asterisk.org/r/1414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334514 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5bf2fe764a |
Merged revisions 334453 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines Make SQL query in app_voicemail.c portable LIMIT is not portable. Regression from r312212 (closes issue ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1415/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334455 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2ce6c668ca |
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/branches/10@334357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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57acdddb2d |
Merged revisions 334296 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011) | 39 lines Fix potential memory allocation failure crashes in config.c. * Added required checks to the returned memory allocation pointers to prevent crashes. * Made ast_include_rename() create a replacement ast_variable list node if the new filename is longer than the available space. Fixes potential crash and memory leak. * Factored out ast_variable_move() from ast_variable_update() so ast_include_rename() can also use it when creating a replacement ast_variable list node. * Made the filename stuffed at the end of the struct a minimum allocated size in ast_variable_new() in case ast_include_rename() changes the stored filename. * Constify struct char pointers pointing to strings stuffed at the end of the struct for: ast_variable, cache_file_mtime, and ast_config_map. * Factored out cfmtime_new() to remove inlined code and allow some struct pointers to become const. * Removed the list lock from struct cache_file_mtime that was never used. * Added doxygen comments to several structure elements and better documented what strings are stuffed at the struct end char array. * Reworked ast_config_text_file_save() and set_fn() to handle allocation failure of the include file scratch pad object tracking blank lines. * Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure it is long enough for any filename with path. Also reduced the number of container fileset buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review: https://reviewboard.asterisk.org/r/1378/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334297 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 334234 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01 Sep 2011) | 2 lines Remove 1.6 compatibility documentation from 1.8, as it no longer applies. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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26ab0f9af9 |
fixed a typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334233 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a5922759ce |
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/branches/10@334230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 334156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug 2011) | 4 lines Disable T.38 when we get a invite with image media port set to 0 ASTERISK-17678 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334157 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c4cdaf31c6 |
only alter the gateway_timeout when attching the gateway to a channel
ASTERISK-18219 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334064 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2723aba2fa |
Merged revisions 334012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) | 23 lines No DAHDI channel available for conference, user introduction disabled. The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards Patches: jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334013 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e6de0ecf48 |
Merged revisions 334009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines Call pickup race leaves orphaned channels or crashes. Multiple users attempting to pickup a call that has been forked to multiple extensions either crashes or fails a masquerade with a "bad things may happen" message. This is the scenario that is causing all the grief: 1) Pickup target is selected 2) target is marked as being picked up in ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app dial or queue gets a chance to hang up losing calls and calls ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with ast_channel_masquerade(), ast_hangup() completes successfully and the channel is no longer in the channels container. 6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the masquerade on the dead channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel 8) bad things happen while doing the masquerade and in the process ast_do_masquerade() puts the dead channel back into the channels container 9) The "orphaned" channel is visible in the channels list if a crash does not happen. This patch does the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel and not release the channel lock until that has happened. * Made __ast_channel_masquerade() not setup a masquerade if either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work. (closes issue ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer Review: https://reviewboard.asterisk.org/r/1400/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1180e8d803 |
Merged revisions 334006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines Correct an AMI protocol violation with SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are ended by using \r\n this confuses any interfacing script. (closes issue ASTERISK-17486) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334007 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7adad5c8ef |
security fix. really drop call if signalling addr is not same as socket
addr git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e565b2c880 |
Merged revisions 333947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 lines cleanups in ACF/ARJ GK replies processing fixed long (24 sec) pause if acf/arj proccessed before ast_cond_wait called to wait this ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333961 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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12e0da4667 |
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/branches/10@333895 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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02d2a80332 |
Merged revisions 333836 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) | 15 lines Refresh peer address if DNS unavailable at peer creation If Asterisk starts and no DNS is available, outbound registrations will fail indefinitely. This patch copies the address from the sip_registry struct, which will be updated, to the peer->addr when necessary. If dnsmgr is enabled, the registration fails without the patch because even though the address on the registry is updated via dnsmgr, the address is just copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update the address that is copied to the sip_pvt or peers. Closes issue ASTERISK-18000 Review: https://reviewboard.asterisk.org/r/1335/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333837 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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dc7d36333c |
Merged revisions 333784-333785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011) | 2 lines Fix deadlock potential of chan_mobile.c:mbl_ast_hangup(). ........ r333785 | rmudgett | 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line Add some do not hold locks notes to channel.h ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333786 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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13288c9c5d |
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/branches/10@333716 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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90c4666e77 |
Use realtime text when it is negotiated
This patch make use of wirte_text() realtime text instead of send_text() if T.140 is in native formats. ASTERISK-17937 Review: https://reviewboard.asterisk.org/r/1356/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333681 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 333630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29 Aug 2011) | 1 line Fixed improperly formatted TestEvent AMI message in app_voicemail ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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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/branches/10@333570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a087c5c739 |
chan_vpb: remove unused variables (gcc4.6)
GCC 4.6 detects variables that get assined to, but never used later. Also removes some remmed-out lines that become invalid. (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>, git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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43dee65979 |
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/branches/10@333410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 333339 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011) | 20 lines Bug fixes for voicemail user emailsubject / emailbody. This code change fixes a few issues with the voicemail user override of emailbody and emailsubject, including escaping the strings, potential memory leaks, and not overriding the voicemail defaults. Revision 325877 fixed this for ASTERISK-16795, but did not fix it for ASTERISK-16781. A subsequent check-in prevented 325877 from being applied to 10. This check-in resolves both issues, and applies the changes to 1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by: Sebastien Couture Tested by: mjordan (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1374 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333370 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 333267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug 2011) | 2 lines Fix for DESTDIR spaces patch. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 333265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) | 14 lines Segfault when publishing device states via XMPP and not connected When using publishing device state with res_jabber, Asterisk will attempt to send a device state using the unconnected client using iks_send_raw and crash. This patch checks the validity of the connection before attempting to send the device state. (closes issue ASTERISK-18078) Reported by: Michael L. Young Patches: res_jabber-segfault-pubsub-not-connected2.patch (license #5026) patch uploaded by Michael L. Young Tested by: Jonathan Rose ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 333201 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines Fix installation into directories containing spaces. This also vastly simplifies the logic in sounds/Makefile (Closes issue ASTERISK-18290) Reported by: Paul Belanger Review: https://reviewboard.asterisk.org/r/1379/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333203 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a0c2e7ac31 |
Changed the "timeout" option to "gwtimeout".
ASTERISK-18219 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333115 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1a85bf60a2 |
Merged revisions 333010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011) | 12 lines Memory Leak in app_queue The patch that was committed in the 1.6.x versions of Asterisk for ASTERISK-15862 actually fixed two issues. One was not applicable to 1.8 but the other is. queue_leak.patch fixes the portion applicable to 1.8. (closes issue ASTERISK-18265) Reported by: Fred Schroeder Patches: queue_leak.patch (license #5049) patch uploaded by mmichelson Tested by: Thomas Arimont ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333011 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 332939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011) | 7 lines Minor code optimizations. * Simplify ast_category_browse() logic for easier understanding. * Remove dead code in ast_variable_delete() and simplify some of its logic. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d3082a6e01 |
Merged revisions 332874 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011) | 18 lines Reference leaks in app_queue. * Fixed load_realtime_queue() leaking a queue reference when it overwrites q when processing a realtime queue. (issue ASTERISK-18265) * Make join_queue() unreference the queue returned by load_realtime_queue() when it is done with the pointer. The load_realtime_queue() returns a reference to the just loaded realtime queue. * Fixed queues container reference leak in queues_data_provider_get(). * queue_unref() should not return q that was just unreferenced. * Made logic in __queues_show() and queues_data_provider_get() when calling load_realtime_queue() easier to understand. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332878 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 332876 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon, 22 Aug 2011) | 6 lines Revert previous commit It seems google is still making changes to the protocol. (issue ASTERISK-18301) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4731db7a16 |
Fix merge property.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332875 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 332817 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) | 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332832 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 332816 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011) | 8 lines Memory leaks in realtime_multi_xxx() when database access returns error. * Fix realtime_multi_pgsql() configuration memory leak when the database access returns an error. * Fix realtime_multi_odbc() configuration category use after free when the database access returns an error. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332830 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 332759 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011) | 15 lines Memory leak reading realtime database variable list. Calling ast_load_realtime() can leak the last list node if the read list only contains empty variable value items. * Fixed list filter loop in ast_load_realtime() to delete the list node immediately instead of the next time through the loop. The next time through the loop may not happen if the node to delete is the last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265) Patches: jira_asterisk_18265_v1.8_config.patch (license #5621) patch uploaded by rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332761 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f3c6244fa9 |
add a way to disable and/or modify the gateway timeout
ASTERISK-18219 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 332699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun, 21 Aug 2011) | 5 lines Fix outgoing calls in chan_gtalk (closes issue ASTERISK-18301) Reported by: az1324 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |