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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
2010-10-08Merged revisions 290863 via svnmerge from jpeeler1-4/+5
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: http://svn.digium.com/svn/asterisk/branches/1.8@290864 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-07Add Philippe Sultan to chan_gtalk author list.dvossel1-0/+2
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: http://svn.digium.com/svn/asterisk/branches/1.8@290829 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-07Outbound gtalk calls now work correctly.dvossel1-1/+1
There was a problem with how the candidates were being built on an outbound call. This patch fixes that. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290828 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-07Merged revisions 290751 via svnmerge from qwell3-32784/+7724
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: http://svn.digium.com/svn/asterisk/branches/1.8@290752 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-07Merged revisions 290712 via svnmerge from russell1-2/+9
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: http://svn.digium.com/svn/asterisk/branches/1.8@290713 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-06Fixes commented out code to use #if 0 instead.dvossel1-4/+6
Thanks to rmudgett for catching this! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290674 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-06Fixes gtalk outbound DTMF to work properly.dvossel1-8/+38
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: http://svn.digium.com/svn/asterisk/branches/1.8@290648 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-06Merged revision 290613 fromrmudgett1-2/+1
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: http://svn.digium.com/svn/asterisk/branches/1.8@290614 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-06Merged revisions 290575 via svnmerge from tilghman1-3/+5
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: http://svn.digium.com/svn/asterisk/branches/1.8@290576 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-06Don't try to send RTP when remote_address is nulltwilson1-0/+5
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: http://svn.digium.com/svn/asterisk/branches/1.8@290542 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Fixes uninitialized memory problem in 'iax2 set debug peer' option.dvossel1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290506 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Fixes chan_gtalk to work with gmail clientdvossel4-97/+182
This patch was written by Philippe Sultan (phsultan). Thanks for keeping this up to date! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290479 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Merged revisions 290396 via svnmerge from tilghman1-5/+8
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500 (Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010) | 8 lines Fix a crash by ensuring that we don't alter memory after it's freed. (closes issue #17387) Reported by: jmls Patches: 20100726__issue17387.diff.txt uploaded by tilghman (license 14) Tested by: jmls ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290408 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Resolves dnsmgr memory corruption in chan_iax2.dvossel1-141/+190
(closes issue #17902) Reported by: afried Patches: issue_17902.rev1.txt uploaded by russell (license 2) Tested by: afried, russell, dvossel Review: https://reviewboard.asterisk.org/r/965/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290378 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Merged revisions 290375 via svnmerge from dvossel1-11/+17
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010) | 10 lines Fixes PickupChan() not working with full channel name. (closes issue #18011) Reported by: schern Patches: app_directed_pickup.c.2.patch uploaded by schern (license 995) app_directed_pickup.c.trunk.patch uploaded by schern (license 995) Tested by: schern, dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290376 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Restore run directory for OS X, as well as standardizing some other paths to ↵tilghman2-1/+16
Mac OS X. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290289 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-04Merged revisions 290254 via svnmerge from tilghman11-1846/+1929
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010) | 11 lines Change new pattern matcher to regard dashes the same as the old pattern matcher -- as visual candy to be ignored. Also change the AEL parser to not generate dashes within extensions, as those dashes would be ignored. Update the AEL tests to match this behavior. (closes issue #17366) Reported by: murf Patches: 20100727__issue17366.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290255 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-04Merged revisions 290201 via svnmerge from tilghman2-15/+15
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500 (Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04 Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290209 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-03Merged revisions 290101 via svnmerge from tilghman2-6/+166
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500 (Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03 Oct 2010) | 2 lines Automatically re-run configure test for menuselect, when the relevant makeopts settings change. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290102 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-03Get notification only when file is closed, not when created.tilghman1-1/+1
(closes issue #17924) Reported by: mkeuter Patches: asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946) Tested by: abelbeck git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290066 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-02Allow users to pass additional arguments to the Subversion command thatkpfleming1-1/+1
obtains the MP-3 source code. (reported on IRC by jmls) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290026 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-02Merged revisions 289950 via svnmerge from oej1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör, 02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2 lines Add documentation for undocumented option to AMI action originate ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289951 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-02Merged revisions 289874 via svnmerge from tilghman1-12/+6
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500 (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) | 8 lines When forwarding a message, a prepend means that the filesystem will always have a better copy. (closes issue #17803) Reported by: dpetersen Patches: 20100923__issue17803.diff.txt uploaded by tilghman (license 14) Tested by: dpetersen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289875 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-02Merged revisions 289798 via svnmerge from jpeeler4-2/+21
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289840 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289704 via svnmerge from pabelanger2-41/+36
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400 (Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct 2010) | 6 lines Disable debugging by default and reformat .config file. Review: https://reviewboard.asterisk.org/r/929/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289718 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289700 via svnmerge from jpeeler1-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500 (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines Ensure user portion of SIP URI matches dialplan when using encoded characters. This commit takes a simliar approach to 288112 and checks the dialplan to determine the proper action for an incoming contact header as to whether or not it should be decoded or not. sip_new was blindly always decoding the extension, which also caused the outgoing contact header to be incorrect as well as failing to match the encoded extension in the dialplan. (closes issue #17892) Reported by: wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license 325) Tested by: wdoekes ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289701 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01don't iterate through all dialogs to find and delete old subscribesschmitds1-36/+5
On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed. (closes issue #17950) Reported by: schmidts Tested by: schmidts Review: https://reviewboard.asterisk.org/r/901/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289622 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-30Solaris fixes.tilghman1-1/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289581 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-30Merged revisions 289553 via svnmerge from mnicholson1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep 2010) | 4 lines Properly handle channel allocation failures duing invites with replaces. ABE-2588 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289554 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-30Merged revision 289547 fromrmudgett1-1/+14
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu, 30 Sep 2010) | 10 lines In chan_misdn, the DivertingLegInformation2 DivertingNr is garbage when the number is restricted. The same thing happens with DivertingLegInformation1 DivertedTo number. The misdn_PresentedNumberUnscreened_extract() extracted the Unscreened PartyNumber field unconditionally. It now checks the presented number unscreened type to see if the PartyNumber was even present. JIRA ABE-2595 .......... git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289549 f38db490-d61c-443f-a65b-d21fe96a405b
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