mirror of https://github.com/asterisk/asterisk
master
20
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${ noResults }
20643 Commits (be6ddcedcd0c79f81a3bccca11970efd008fa213)
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be6ddcedcd |
Merged revisions 292224 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500 (Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010) | 9 lines Add support for the new English (Australian Accent) sound files. (closes issue #17426) Reported by: camsown Patches: core-sounds-en_AU.txt uploaded by camsown (license 1050) add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested by: camsown, lmadsen, jtodd, qwell ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292225 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f10b21bd24 |
Resolve some compiler errors in ast_sockaddr_is_any().
These errors came up once this function was used from within netsock2.c.
The errors were like the following:
netsock2.c:393: error: dereferencing pointer ‘({anonymous})’ does break strict-aliasing rules
The usage of a union here avoids this problem.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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16 years ago |
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64a0eebfa3 |
Fixes build error for systems not supporting IPV6_TCLASS.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292155 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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aa0351aac9 |
Fix the cmgr parser.
(closes issue 0018152) Reported by: menschentier git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292122 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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df369754da |
Fixes qos settings for sockets bound to any IPv6 or IPv4 address.
(closes issue #18099) Reported by: jamesnet Patches: issues_18099_v3.diff uploaded by dvossel (license 671 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292085 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d04d58c7c6 |
Disable use of inotify for call file handling as it is not working properly.
(related to #18089) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292083 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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12211219dc |
Merged revisions 292049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) | 15 lines Base directory for MOH should be ASTDATADIR If the directive 'directory' is relative, make it relative to the datadir, rather than to the varlibdir. In the sample configuration it is relative ('moh'). This has no effect unless you have actively set the datadir explicitly (at build time or at run time). (closes issue #16906) Patches: moh_datadir uploaded by tzafrir (license 46) Review: https://reviewboard.asterisk.org/r/974/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292050 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5b5faf3fc5 |
Ref/unref res_srtp when we create/destroy a session
This avoids unhappy crashing when we try to 'core stop gracefully' and res_srtp tries to unload before chan_sip does. Thanks, Russell! (closes issue #18085) Reported by: st git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292016 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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3e3ea54864 |
Fixes peer's host port information being lost on sip reload.
(closes issue #18135) Reported by: lmadsen Patches: crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291942 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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67f8391cee |
Merged revisions 291939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400 (Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, 15 Oct 2010) | 2 lines Clean up formatting. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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36a672b7e8 |
Merged revisions 291904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010) | 7 lines Don't crash or deadlock on module unload We can't hold the lock while pthread_join is called since aji_log_hook will attempt to lock from the other therad. We reorder the pthread_join and ast_aji_disconnect so that we don't do an SSL_read() while SSL_shutdown is running, causing a crash. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e735b3f28c |
Set TCLASS field of IPv6 header when sip qos options are set.
(closes issue #18099) Reported by: jamesnet Patches: issues_18099_v2.diff uploaded by dvossel (license 671) Tested by: dvossel, jamesnet git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291829 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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24941a72ad |
Safer xml parsing, treat all clients the same, and better local candidate selection.
The gtalk channel driver was doing several unsafe operations in regards to how it parsed incoming XML messages. I have cleaned that code up so it should be much safer now. We now treat all clients types the same. We have no reason to distinguish between GMAIL and GOOGLE VOICE clients anymore because they all work the same way. I also modified how the local ip is found. If no bindaddress is provided in the config file, we attempt to determine the local ip we would use to connect to google.com. If that fails, then we fall back to the ast_find_ourip() function as a last resort. Using the new method makes it much less likely that we would ever advertise a local RTP candidate as a loopback address. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291827 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c55bdb40f0 |
Add missing ifdefs for test framework and new locale code.
(closes issue #18137) Reported by: ovi Patches: 18137_test_framework_ifdef.patch uploaded by wdoekes (license 717) 18137_localelist_warning.patch uploaded by wdoekes (license 717) Tested by: ovi git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291791 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a37956721c |
Add the ability for ast_find_ourip to return IPv4, IPv6 or both.
While testing chan_gtalk I noticed jabber was using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip() to return both IPv6 and IPv4 results. Adding a family parameter gives you the ablility to choose. Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results. Review: https://reviewboard.asterisk.org/r/973/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291758 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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27810a962e |
Fix a typo - s/seucre/secure/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291725 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b8f40f4805 |
Merged revisions 291655 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500 (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010) | 20 lines Deadlock between dahdi_exception() and dahdi_indicate(). There is a deadlock between dahdi_exception() and dahdi_indicate() for analog ports. The call-waiting and three-way-calling feature can experience deadlock if these features are trying to do something and an event from the bridged channel happens at the same time. Deadlock avoidance code added to obtain necessary channel locks before attemting an operation with call-waiting and three-way-calling. (closes issue #16847) Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/971/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291656 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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9b9204c5ae |
Merged revisions 291580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291580 | twilson | 2010-10-13 15:58:43 -0700 (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) | 21 lines Don't ignore frames that have been queued when softhangup'd When an outgoing call is answered and hung up by the far end *very* quickly, we may not read any frames and therefor end up with a call that displays the wrong disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately sets the _softhangup flag on the channel and then queues the HANGUP control frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates that a hangup request has been made (which it will if _softhangup is set). So, we end up losing control frames. This change makes __ast_read continue to read frames even if a soft hangup has been requested. It queues a hangup frame to make sure that __ast_read() will still eventually return NULL. Much thanks to David Vossel for all of the reviews, discussion, and help! (closes issue #16946) Reported by: davidw Review: https://reviewboard.asterisk.org/r/740/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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912ead1a24 |
More fixup for chan_gtalk.
This patch makes the xml parsing safer. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291578 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e6fb66732f |
Add a simple AMI client web page
This patch uses the XML docs to parse all of the available AMI commands and allows you to enter the command name and be presented with a form with the available fields. You can then rapidly tab through the fields and submit the command and view the response. It is much faster/easier than having to use telnet for testing purposes. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d8225d08b9 |
The chan_dahdi faxdetect option only works for the first FAX call.
The chan_dahdi faxdetect option only works for the first call. After that the option no longer works. The struct dahdi_pvt.callprogress member is the encoded user config setting for the callprogress and faxdetect config options. Changing this value alters the configuration for all following calls until the chan_dahdi.conf file is reloaded. * Fixed the chan_dahdi ast_channel_setoption callback to not change the users faxdetect config setting except for the current call. * Fixed the chan_dahdi ast_channel_queryoption callback to read the active DSP setting of the faxdetect option. * Made actually disable the active faxdetect DSP setting for the current call on the analog port. my_handle_dtmfup() is used for normal analog ports. dahdi_handle_dtmfup() is the legacy code and is no longer used unless in a radio mode. (closes issue #18116) Reported by: seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/972/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291541 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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84b12f0c4d |
Merged revision 291504 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed, 13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the ast_channel. Must get the ast_channel lock before proceeding with release_chan() and release_chan_early() to hold off ast_hangup() from destroying the ast_channel. Missed this change for -r291468. JIRA ABE-2598 JIRA SWP-2317 .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a5a798aefa |
Merge revision 291468 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed, 13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE --> RELEASE_COMPLETE * Add lock protection around channel list for find/add/delete operations. * Protect misdn_hangup() from release_chan() and vise versa using the release_lock. JIRA ABE-2598 JIRA SWP-2317 .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ec05b242dd |
Merged revisions 291393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291393 | russell | 2010-10-13 10:29:21 -0500 (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines Lock pvt so pvt->owner can't disappear when queueing up a frame. This fixes a crash due to a hangup race condition. ABE-2601 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291394 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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379cf8ce1e |
Merged revisions 291280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010) | 7 lines Add undocumented variables to phoneprov.conf.sample (closes issue #18107) Reported by: lathama Patches: phoneprov.conf.sample.diff uploaded by lathama (license 1028) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291284 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c4ead286ef |
Merged revisions 291264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291264 | tilghman | 2010-10-12 12:05:31 -0500 (Tue, 12 Oct 2010) | 9 lines Merged revisions 291263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010) | 2 lines Oops, incorrect range (although unallocated at ARIN) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5503797885 |
Merged revisions 291229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010) | 2 lines Add documention that mentions options are defined but not used. (Issue #18101) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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197b033cd7 |
Fixes manager.c crash.
This issue was caused by improper use of the mansession lock and manession_session lock. These two structures are confusing to begin with so I'm not surprised this occurred. I fixed this by consistently making sure we use each of these locks only to protect the data in the corresponding structure. We had mismatched usage of these locks which resulted in no mutual exclusivity occurring at all. (closes issue #17994) Reported by: vrban Patches: mansession_locking_fix.diff uploaded by dvossel (license 671) Tested by: vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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397a910b5c |
Update CHANGES to reflect new gtalk.conf options.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291194 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f5782280d8 |
Gtalk enhancements and general code cleanup.
This patch includes several chan_gtalk enhancements. Two new gtalk.conf options have been added, externip and stunadd. Setting externip allows us to manually specify what the external IP address is outside of a NAT environment. Setting the stunaddr option to a valid stun server allows for that external ip to be retrieved via a STUN server automatically. This external IP is then advertised during call setup as a possible candidate. I have also attempted to clean up chan_gtalk's code so it meets our coding guidelines. During this cleanup I noticed several things that need to be done in the code and made a TODO section at the top of the file. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291192 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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184d0e7f1b |
Move declaration closer to where now used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a96796cc44 |
Merged revisions 291110-291111 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500 (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line Add missing unlock to an exception condition in reload_config(). ........ ................ r291111 | rmudgett | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit from handle_request_do() consistent. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d6214fe55d |
Merged revisions 291073 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010) | 15 lines Fixed infinite loop in verbose/debug message output. Setting the module/filename specific message level and then changing it resulted in the linked list being looped on itself. Traversing this linked list is an infinite loop if what you are looking for is not in the list. Also plugged some CLI parsing holes in the associated CLI command: * Removing a nonexistent module from the list actually added it with a level of zero. * Setting the non-module specific level to zero is now equivalent to setting it to "off" as documented. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291075 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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388afd2b80 |
Add missing option to set calls to be logged in GMT/UTC.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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495b8f3719 |
small correction for verbose print h.323 packets
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291037 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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409feba665 |
Added fast start and h.245 tunneling options per user and peer.
Added options for faststart/h.245 tunneling per user/peer, properly handle these and global options, correction of handling fs/tunneling fields in signalling responses (issue #17972) Reported by: salecha Patches: fs-tunnel-per-point-3.patch uploaded by may213 (license 454) Tested by: may213, salecha git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6412f8d665 |
Make outbound Google Voice calls.
This patch allows for outbound Google Voice calls to be dialed from Asterisk using chan_gtalk. Below is an example dialstring. exten -> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,) In this example, 'asterisk' is the jabber.conf profile configured to connect to your gmail account. In order to receive Google Voice calls make sure to enable 'allowguest=yes' in gtalk.conf. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290973 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6f18d825b2 |
Parentheses around assignment used as truth value, introduced in r290937.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290938 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0bece542ff |
Add option to res_config_mysql and app_mysql to specify a character set that MySQL should use.
(closes issue 17948) Reported by qmax. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290937 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c55ebead20 |
Merged revisions 290863 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500 (Thu, 07 Oct 2010) | 16 lines Merged revisions 290862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed at control console. A recent change was made to avoid a race condition on shutdown which only called the end functions from the console thread. However, when pressing Ctrl-C the quit handler is called from the signal handler thread. (closes issue #17698) Reported by: jmls ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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45a6b95d93 |
Add Philippe Sultan to chan_gtalk author list.
Philippe has made some notable contributions to the gtalk channel driver. His name deserves to be listed amoung the authors of that file. Thanks Philippe! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290829 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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90be613635 |
Outbound gtalk calls now work correctly.
There was a problem with how the candidates were being built on an outbound call. This patch fixes that. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290828 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c437bca01b |
Merged revisions 290751 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290751 | qwell | 2010-10-07 15:57:14 -0500 (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | 9 lines Allow PRI to build properly when using --with-pri. Use the directories found for the parent when using lib dependencies. (closes issue #17314) Reported by: tzafrir Patches: 17314-withdeps.diff uploaded by qwell (license 4) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290752 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b61babaf58 |
Merged revisions 290712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010) | 4 lines Don't crash when Set() is called without a value. Review: https://reviewboard.asterisk.org/r/949/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290713 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7853c1cd2f |
Fixes commented out code to use #if 0 instead.
Thanks to rmudgett for catching this! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290674 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1d19ab1a63 |
Fixes gtalk outbound DTMF to work properly.
Outbound DTMF with gtalk needs to be done within the RTP stream. I discovered this after investigating a packet capture from the gmail client. Instead of performing jingle signaling DTMF, the gtalk servers expect all DTMF to arrive on the RTP stream using RFC2833 way of doing things. Chan_gtalk also had an issue with negotiating RTP payload type 106 for the telephony-event and then sending DTMF as payload 101. This has been resolved by always negotiating 101 as the payload type like we do everywhere else. With this patch, incoming google voice calls forwarded to Asterisk via gtalk work. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290648 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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966c392632 |
Merged revision 290613 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed, 06 Oct 2010) | 5 lines Eliminate a redundant test for AST_CONTROL_REDIRECTING. Eliminate redundant test for AST_CONTROL_REDIRECTING that prevents running the redirecting interception macro if it is defined. .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290614 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b112c3de5c |
Merged revisions 290575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010) | 8 lines Allow streaming audio from a pipe. (closes issue #18001) Reported by: jamicque Patches: 20100926__issue18001.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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13562fd2e3 |
Don't try to send RTP when remote_address is null
It is possible for ast_rtp_stop() to be called which will clear the remote address and cause the sendto to fail and spam warnings. Don't send in this case. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cb0e2d3bd7 |
Fixes uninitialized memory problem in 'iax2 set debug peer' option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290506 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |