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2010-10-21Importing release summary for 1.8.0 release.v1.8.0lmadsen2-0/+24689
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0@292594 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-21Drop incorrect summary file.lmadsen2-5045/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0@292588 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-21Importing release summary for 1.8.0 release.lmadsen2-0/+5045
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0@292587 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-21Update .version and ChangeLog.lmadsen4-207/+5
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0@292586 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-21Create 1.8.0 from 1.8.0-rc5.lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0@292580 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Use autotagged externalsv1.8.0-rc5lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0-rc5@292278 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Importing release summary for 1.8.0-rc5 release.lmadsen2-0/+206
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0-rc5@292276 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Importing files for 1.8.0-rc5 release.lmadsen3-0/+25504
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0-rc5@292275 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Creating tag for the release of asterisk-1.8.0-rc5lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0-rc5@292261 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Merged revisions 292229 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292229 | lmadsen | 2010-10-18 17:01:16 -0500 (Mon, 18 Oct 2010) | 3 lines Fix typo in the sounds/Makefile. (Issue #17426) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292230 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Merged revisions 292226 via svnmerge from jpeeler1-2/+12
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500 (Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) | 11 lines Fix improper operator key acceptance and clean up temp recording files. This is a fix for when pressing the operator key after recording an unavailable, busy, name, or temporary message in mailbox options. The operator key should not be accepted here, but should be allowed during the message recording. If the operator key is pressed during ensure the file is saved or deleted as apporopriate. Also, ensure removal of temporary recorded files after an early hang up or when message acceptance confirmation times out. ABE-2518 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292227 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Merged revisions 292224 via svnmerge from lmadsen2-4/+25
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: http://svn.digium.com/svn/asterisk/branches/1.8@292225 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Resolve some compiler errors in ast_sockaddr_is_any().russell1-5/+10
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: http://svn.digium.com/svn/asterisk/branches/1.8@292188 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Fixes build error for systems not supporting IPV6_TCLASS.dvossel1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292155 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Fix the cmgr parser.mnicholson1-1/+1
(closes issue 0018152) Reported by: menschentier git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292122 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Fixes qos settings for sockets bound to any IPv6 or IPv4 address.dvossel1-11/+24
(closes issue #18099) Reported by: jamesnet Patches: issues_18099_v3.diff uploaded by dvossel (license 671 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292085 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Disable use of inotify for call file handling as it is not working properly.jpeeler1-0/+5
(related to #18089) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292083 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-16Merged revisions 292049 via svnmerge from tzafrir2-2/+2
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: http://svn.digium.com/svn/asterisk/branches/1.8@292050 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-15Ref/unref res_srtp when we create/destroy a sessiontwilson1-0/+2
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: http://svn.digium.com/svn/asterisk/branches/1.8@292016 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-15Fixes peer's host port information being lost on sip reload.dvossel1-0/+3
(closes issue #18135) Reported by: lmadsen Patches: crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by: lmadsen git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291942 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-15Merged revisions 291939 via svnmerge from pabelanger1-16/+13
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: http://svn.digium.com/svn/asterisk/branches/1.8@291940 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-15Merged revisions 291904 via svnmerge from twilson1-3/+3
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: http://svn.digium.com/svn/asterisk/branches/1.8@291905 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-14Set TCLASS field of IPv6 header when sip qos options are set.dvossel1-1/+10
(closes issue #18099) Reported by: jamesnet Patches: issues_18099_v2.diff uploaded by dvossel (license 671) Tested by: dvossel, jamesnet git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291829 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-14Safer xml parsing, treat all clients the same, and better local candidate ↵dvossel1-133/+92
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: http://svn.digium.com/svn/asterisk/branches/1.8@291827 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-14Add missing ifdefs for test framework and new locale code.jpeeler1-1/+6
(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: http://svn.digium.com/svn/asterisk/branches/1.8@291791 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-14Add the ability for ast_find_ourip to return IPv4, IPv6 or both.pabelanger6-11/+15
While testing chan_gtalk I noticed jabber was using my IPv6 address and not IPv4. When using bindaddr= 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: http://svn.digium.com/svn/asterisk/branches/1.8@291758 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-14Fix a typo - s/seucre/secure/russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291725 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291655 via svnmerge from rmudgett3-94/+274
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: http://svn.digium.com/svn/asterisk/branches/1.8@291656 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291580 via svnmerge from twilson1-1/+12
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: http://svn.digium.com/svn/asterisk/branches/1.8@291581 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13More fixup for chan_gtalk.dvossel1-42/+64
This patch makes the xml parsing safer. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291578 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Add a simple AMI client web pagetwilson2-1/+179
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: http://svn.digium.com/svn/asterisk/branches/1.8@291575 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13The chan_dahdi faxdetect option only works for the first FAX call.rmudgett1-7/+11
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: http://svn.digium.com/svn/asterisk/branches/1.8@291541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revision 291504 fromrmudgett1-35/+49
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: http://svn.digium.com/svn/asterisk/branches/1.8@291507 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merge revision 291468 fromrmudgett1-72/+151
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: http://svn.digium.com/svn/asterisk/branches/1.8@291469 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291393 via svnmerge from russell1-0/+16
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: http://svn.digium.com/svn/asterisk/branches/1.8@291394 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-12Merged revisions 291280 via svnmerge from lmadsen1-0/+14
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: http://svn.digium.com/svn/asterisk/branches/1.8@291284 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-12Merged revisions 291264 via svnmerge from tilghman1-2/+2
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: http://svn.digium.com/svn/asterisk/branches/1.8@291265 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-12Merged revisions 291229 via svnmerge from lmadsen1-2/+2
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: http://svn.digium.com/svn/asterisk/branches/1.8@291230 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-12Fixes manager.c crash.dvossel1-16/+16
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: http://svn.digium.com/svn/asterisk/branches/1.8@291227 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Update CHANGES to reflect new gtalk.conf options.dvossel1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291194 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Gtalk enhancements and general code cleanup.dvossel4-144/+146
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: http://svn.digium.com/svn/asterisk/branches/1.8@291192 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Move declaration closer to where now used.rmudgett1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291113 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Merged revisions 291110-291111 via svnmerge from rmudgett1-4/+9
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: http://svn.digium.com/svn/asterisk/branches/1.8@291112 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Merged revisions 291073 via svnmerge from rmudgett1-17/+39
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: http://svn.digium.com/svn/asterisk/branches/1.8@291075 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-09Add missing option to set calls to be logged in GMT/UTC.tilghman2-10/+38
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291038 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-09small correction for verbose print h.323 packetsmay1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291037 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-09Added fast start and h.245 tunneling options per user and peer.may3-37/+107
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: http://svn.digium.com/svn/asterisk/branches/1.8@291005 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-08Make outbound Google Voice calls.dvossel1-3/+15
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: http://svn.digium.com/svn/asterisk/branches/1.8@290973 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-08Parentheses around assignment used as truth value, introduced in r290937.espiceland1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290938 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-08Add option to res_config_mysql and app_mysql to specify a character set that ↵espiceland3-6/+34
MySQL should use. (closes issue 17948) Reported by qmax. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290937 f38db490-d61c-443f-a65b-d21fe96a405b