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2009-03-13Blocked revisions 182071 via svnmergemmichelson0-0/+0
........ r182071 | mmichelson | 2009-03-13 12:49:01 -0500 (Fri, 13 Mar 2009) | 3 lines Remove ast_ prefix from functions which are not public. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182072 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-13Merged revisions 182029 via svnmerge from mmichelson1-6/+16
https://origsvn.digium.com/svn/asterisk/trunk ................ r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar 2009) | 41 lines Merged revisions 181990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182064 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-13Merged revisions 182022 via svnmerge from file1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 | file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix an issue with requesting a T38 reinvite before the call is answered. The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182036 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-13Blocked revisions 181985 via svnmergekpfleming0-0/+0
........ r181985 | kpfleming | 2009-03-13 11:55:38 -0500 (Fri, 13 Mar 2009) | 1 line improve a bit of suboptimal code ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181986 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181846 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 | mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 lines Run the macro on the queue member's channel when he answers, not the caller's channel. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181848 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181769 via svnmerge from mmichelson1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar 2009) | 28 lines Merged revisions 181768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181770 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181731 via svnmerge from tilghman2-9/+18
https://origsvn.digium.com/svn/asterisk/trunk ........ r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines Adjust translation table column widths based upon the translation times. Previously, only 5 columns were displayed, and if a translation time exceeded 99,999 useconds, it would be displayed as 0, instead of its actual time. (closes issue #14532) Reported by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman (license 14) Tested by: pj ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181732 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181665 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu, 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 lines Fix incorrect usage of strncasecmp... I really meant to use strcasecmp. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181666 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181661 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines Fix another scenario where depending on configuration the stream would not get read. For custom commands we don't know whether the audio is coming from a stream or not so we are going to have to read the data despite no channels. (closes issue #14416) Reported by: caspy ........ r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in previous commit. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181662 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181656 via svnmerge from file1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines Fix issue with streaming MOH failing if nobody is listening. When a music class is setup to actually provide music on hold from a stream we need to constantly read audio from it since it will constantly be providing audio. This is now done despite there being no channels listening to it. (closes issue #14416) Reported by: caspy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181657 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181612 via svnmerge from file1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 | file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix crash when sleep and retries argument was not given to RetryDial application. (closes issue #14647) Reported by: sherpya ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181613 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181542 via svnmerge from rmudgett1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009) | 1 line Use the correct branch integrated property when generating the version string ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181543 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181499 via svnmerge from mvanbaak1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181509 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Blocked revisions 181465 via svnmergerussell0-0/+0
........ r181465 | russell | 2009-03-11 17:25:57 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGE_PLAY_SOUND variable thread-safe. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181469 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181444 via svnmerge from qwell2-5/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r181444 | qwell | 2009-03-11 17:20:13 -0500 (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | 4 lines Allow prefix to set localstatedir (when used and different from the default). This is similar to the /etc change that was made for the non-FreeBSD case. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181450 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181428 via svnmerge from russell1-8/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGEPVTCALLID variable thread-safe. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181429 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181424 via svnmerge from russell1-14/+31
https://origsvn.digium.com/svn/asterisk/trunk ................ r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181425 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181371 via svnmerge from dvossel2-31/+71
https://origsvn.digium.com/svn/asterisk/trunk ................ r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181372 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181345 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines Merged revisions 181328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181352 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181296 via svnmerge from file1-3/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines Merged revisions 181295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181297 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Blocked revisions 181292 via svnmergerussell0-0/+0
........ r181292 | russell | 2009-03-11 11:19:38 -0500 (Wed, 11 Mar 2009) | 2 lines Replace contents of this doc with a pointer to its new home ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181293 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11add missing header filejpeeler1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181284 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Blocked revisions 181244 via svnmergemmichelson0-0/+0
........ r181244 | mmichelson | 2009-03-11 09:28:40 -0500 (Wed, 11 Mar 2009) | 11 lines Fix segfault when dialing a typo'd queue If trying to dial a non-existent queue, there would be a segfault when attempting to access q->weight, even though q was NULL. This problem was introduced during the queue-reset merge and thus only affects trunk. (closes issue #14643) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181245 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Blocked revisions 181210 via svnmergefile0-0/+0
........ r181210 | file | 2009-03-11 10:44:42 -0300 (Wed, 11 Mar 2009) | 3 lines Don't play the "you are about to be placed into the conference" and "the leader has left the conference" sounds if the quiet option is enabled. (reported by Vadim Lebedev on the asterisk-dev list) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181211 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Fix merge oops from 181137jpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181178 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181135 via svnmerge from jpeeler6-128/+49
https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181137 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Blocked revisions 181099 via svnmergerussell0-0/+0
........ r181099 | russell | 2009-03-10 21:25:24 -0500 (Tue, 10 Mar 2009) | 2 lines tabs to spaces ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181100 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181032-181033 via svnmerge from mmichelson1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500 (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines Remove unused variables. ........ ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500 (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC 3891 ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181034 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Merged revisions 180944 via svnmerge from qwell4-24/+22
https://origsvn.digium.com/svn/asterisk/trunk ................ r180944 | qwell | 2009-03-10 17:03:41 -0500 (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) | 1 line Make things happier when using autoconf 2.62+ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180946 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180942 via svnmergerussell0-0/+0
........ r180942 | russell | 2009-03-10 17:03:16 -0500 (Tue, 10 Mar 2009) | 2 lines Add some notes on getting in contact with the dev community ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180943 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180938 via svnmergerussell0-0/+0
........ r180938 | russell | 2009-03-10 16:55:49 -0500 (Tue, 10 Mar 2009) | 2 lines Remove difficulty and language specifiers ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180939 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180935 via svnmergerussell0-0/+0
........ r180935 | russell | 2009-03-10 16:45:54 -0500 (Tue, 10 Mar 2009) | 2 lines Expand upon documentation of manager event project ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180936 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180898 via svnmergemvanbaak0-0/+0
........ r180898 | mvanbaak | 2009-03-10 22:15:29 +0100 (Tue, 10 Mar 2009) | 10 lines list the move of the astvarrundir from /var/run to /var/run/asterisk (actually its $(localstatedir)/run/asterisk Makes setups with asterisk as non-root easier to manage because you can setup permissions on this dir instead of touching a file and setting permissions on that. Files that come to mind are asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180899 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180862 via svnmergerussell0-0/+0
........ r180862 | russell | 2009-03-10 14:36:21 -0500 (Tue, 10 Mar 2009) | 1 line add more projects ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180863 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180859 via svnmergerussell0-0/+0
........ r180859 | russell | 2009-03-10 14:23:41 -0500 (Tue, 10 Mar 2009) | 1 line add more project ideas ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180860 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Merged revisions 180800 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 | file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines Reset the thread local string buffer when handling the UserEvent action. (closes issue #14593) Reported by: JimDickenson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180801 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10If a port is specified when dialing a peer then use it.file1-0/+5
(closes issue #14626) Reported by: acunningham git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180799 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Blocked revisions 180750 via svnmergerussell0-0/+0
........ r180750 | russell | 2009-03-09 17:00:42 -0500 (Mon, 09 Mar 2009) | 4 lines Add current mentors list, and first pass on a project list broken out of "PineMango" I will work on adding projects that have been sent to be via email tomorrow. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180751 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Blocked revisions 180719 via svnmergejpeeler0-0/+0
........ r180719 | jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines Add Doxygen documentation for API changes from 1.6.0 to 1.6.1 Copied from my review board description: This is a continuation of the API changes documentation started for describing changes between releases. Most of the API changes were pretty simple needing only to be brought to attention via the new "Asterisk API Changes" list. However, if you see anything that needs further explanation feel free to supplement what is there. The current method of documenting is to add (in the header file): \version <ver number> <description of changes> and then to add the function to the change list in doxyref.h on the AstAPIChanges page. I also made sure all the functions that were newly added were tagged with \since 1.6.1. I think this is a good habit to start both for the historical aspect as well as for the future ability to easily add a "New Asterisk API" page. Review: http://reviewboard.digium.com/r/190/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180720 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Ensure that the new outgoing dialog to a peer is able to set the socket ↵file1-7/+11
details, even if the default is present. (closes issue #14480) Reported by: jon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180718 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Blocked revisions 180684 via svnmergerussell0-0/+0
........ r180684 | russell | 2009-03-09 09:14:34 -0500 (Mon, 09 Mar 2009) | 2 lines Add skeleton for GSoC ideas list ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180685 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-07Blocked revisions 180641 via svnmergerussell0-0/+0
........ r180641 | russell | 2009-03-07 09:36:00 -0600 (Sat, 07 Mar 2009) | 7 lines Make some minor updates to the doxygen configuration - add bridges directory to be processed - add some res/ subdirs - alphabetize subdirs - use consistent indentation ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180642 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-06Merged revisions 180579 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600 (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when IMAP storage is enabled. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180582 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-06Merged revisions 180534 via svnmerge from dvossel1-30/+50
https://origsvn.digium.com/svn/asterisk/trunk ................ r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) | 15 lines Merged revisions 180532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) | 9 lines Fix handling of backreferences for ENUM lookups enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180535 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merged revisions 180465 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600 (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts. There was a fix put in a while back so that an X-Asterisk-VM-Context message header was added to stored IMAP voicemails. This would allow for us to differentiate if the same mailbox name was used in multiple contexts. The problem still left was that not all places where messages were retrieved actually attempted to use this header for information when retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain work as expected. (closes issue #13853) Reported by: vicks1 Patches: 13853_v2.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180466 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merged revisions 180383 via svnmerge from mmichelson2-7/+17
https://origsvn.digium.com/svn/asterisk/trunk ................ r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar 2009) | 31 lines Merged revisions 180380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180404 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Blocked revisions 180382 via svnmergemvanbaak0-0/+0
........ r180382 | mvanbaak | 2009-03-05 20:05:20 +0100 (Thu, 05 Mar 2009) | 2 lines Make sure we terminate the first s| command so we can actually produce correct files. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180384 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merged revisions 180373 via svnmerge from kpfleming3-36/+113
https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180377 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Blocked revisions 180369 via svnmergefile0-0/+0
........ r180369 | file | 2009-03-05 14:18:27 -0400 (Thu, 05 Mar 2009) | 13 lines Merge phase 1 support for the new bridging architecture. This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180370 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Blocked revisions 180261 via svnmergerussell0-0/+0
........ r180261 | russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines Resolve object matching issues related to the removal of the sip_user object. Previously, chan_sip had both sip_peer and sip_user objects in memory. A patch went in to remove sip_user to simplify the code, since everything could be done with just sip_peer. This patch resolves some regressions found that were introduced by those changes. This code comes from svn/asterisk/team/group/sip-object-matching/. Here is a list of the changes that have been made: 1) When doing a match by name with the find_peer() function, make it much easier to specify which objects should be matched by having a parameter that specifies exactly which object types should be considered. Also, update find_by_name() to handle this parameter. Finally, update all code to use the new option values. 2) When looking up an object for an outbound request by name, consider peers only. (create_addr()) 3) Only match peers on an incoming registration request. 4) When doing authentication (except for SUBSCRIBE), look up users by name, instead of all objects by name. 5) When doing authentication (except for SUBSCRIBE), after looking for a user by name, look for a peer by IP address, instead of all objects by IP address. 6) When handling the SIP qualify CLI command or manager action, look for a peer by name, instead of any object by name. 7) When handling the SIP unregister CLI command, look for a peer by name, instead of any object by name. 9) In sip_do_debug_peer(), search for a peer by name, instead of any object by name. 9) When handling the SIPPEER() dialplan function, search for a peer by name, instead of any object by name. 10) In the following session timer related functions, st_get_se(), st_get_refresher(), and st_get_mode(), when looking for an object for a given sip_pvt using pvt->peername, look for a peer by name, instead of any object by name. 11) Fix build_peer() to properly handle the case where separate type=peer and type=user entries were specified in sip.conf. (closes issue #14505) Reported by: lmadsen Review: http://reviewboard.digium.com/r/172/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180262 f38db490-d61c-443f-a65b-d21fe96a405b
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