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2009-02-12Merged revisions 175089 via svnmerge from phsultan1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009) | 6 lines Issue a warning message if our candidate's IP is the loopback address. (closes issue #13985) Reported by: jcovert Tested by: phsultan ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175090 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175058 via svnmerge from phsultan1-5/+47
https://origsvn.digium.com/svn/asterisk/trunk ................ r175058 | phsultan | 2009-02-12 11:31:36 +0100 (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175059 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-11Merged revisions 174948 via svnmerge from mmichelson1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) | 35 lines Fix odd "thank you" sound playing behavior in app_queue.c If someone has configured the queue to play an position or holdtime announcement, then it is odd and potentially unexpected to hear a "Thank you for your patience" sound when no position or holdtime was actually announced. This fixes the announcement so that the "thanks" sound is only played in the case that a position or holdtime was actually announced. There is a way that the "thank you" sound can be played without a position or holdtime, and that is to set announce-frequency to a value but keep announce-position and announce-holdtime both turned off. (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch uploaded by putnopvut (license 60) Tested by: caspy ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174949 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-11Merged revisions 174945 via svnmerge from mmichelson6-7/+23
https://origsvn.digium.com/svn/asterisk/trunk ........ r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174946 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-11Blocked revisions 174844 via svnmergefile0-0/+0
........ r174844 | file | 2009-02-11 10:44:47 -0400 (Wed, 11 Feb 2009) | 10 lines Tell the device state core a change happened when a channel is freed but not a specific state. We need to do this because while we know that the freeing of the channel may cause something to become not in use we do not know this for sure. There may be another channel that is still up which would cause it to be in use. (closes issue #13238) Reported by: kowalma Patches: 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174845 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Merged revisions 174805 via svnmerge from mmichelson1-28/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 | mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11 lines Fix potential for stack overflows in app_chanspy.c When using the 'g' or 'e' options, the stack allocations that were used could cause a stack overflow if a spyer stayed on the line long enough without actually successfully spying on anyone. The problem has been corrected by using static buffers and copying the contents of the appropriate strings into them instead of using functions like alloca or ast_strdupa ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174820 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Merged revisions 174764 via svnmerge from mmichelson1-212/+259
https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 | mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21 lines Fix an fd leak that would occur in HTTP AMI sessions The explanation behind this fix is a bit complicated, and I've already typed it up in the code as a huge comment inside of manager.c, so I'll give the abridged version here. We needed a way to separate action-specific data from session-specific data. Unfortunately, the only way to maintain API compatibility and to not have to change every single manager action was to rename the current mansession structure and wrap it inside a new mansession structure which actually contains action- specific data. (closes issue #14364) Reported by: awk Patches: 14364_better.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/148/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174765 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Merged revisions 174710 via svnmerge from file1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 | file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines Only decrease inringing count if above zero. (issue #13238) Reported by: kowalma ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174711 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Merged revisions 174584 via svnmerge from mnicholson1-13/+19
https://origsvn.digium.com/svn/asterisk/trunk ................ r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb 2009) | 25 lines Merged revisions 174583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines Improve behavior of jitterbuffer when maxjitterbuffer is set. This change improves the way the jitterbuffer handles maxjitterbuffer and dramatically reduces the number of frames dropped when maxjitterbuffer is exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all new frames were dropped until the jitterbuffer is empty. This change modifies the code to only drop frames until maxjitterbuffer is no longer exceeded. Also, previously when maxjitterbuffer was exceeded, dropped frames were not tracked causing stats for dropped frames to be incorrect, this change also addresses that problem. (closes issue #14044) Patches: bug14044-1.diff uploaded by mnicholson (license 96) Tested by: mnicholson Review: http://reviewboard.digium.com/r/144/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174596 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Blocked revisions 174580 via svnmergefile0-0/+0
........ r174580 | file | 2009-02-10 13:48:29 -0400 (Tue, 10 Feb 2009) | 4 lines Set the type for the peer structure to be a peer as the default. (closes issue #14447) Reported by: triccyx ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174581 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Merged revisions 174543 via svnmerge from file1-6/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 | file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported. (closes issue #14399) Reported by: caspy (issue #13238) Reported by: kowalma ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174544 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10For some strange reason, I didn't think 1.6.0 neededmurf1-1/+1
this fix. I was wrong. Here it is. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174439 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Blocked revisions 174435 via svnmergemurf0-0/+0
........ r174435 | murf | 2009-02-09 21:49:02 -0700 (Mon, 09 Feb 2009) | 8 lines This patch removes the use of AST_PBX_KEEPALIVE from app_rpt.c. (closes issue #14435) Reported by: D_McNaul ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174436 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Blocked revisions 174432 via svnmergemurf0-0/+0
........ r174432 | murf | 2009-02-09 21:36:22 -0700 (Mon, 09 Feb 2009) | 3 lines More intptr_t work. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174433 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Blocked revisions 174370 via svnmergemurf0-0/+0
................ r174370 | murf | 2009-02-09 19:45:56 -0700 (Mon, 09 Feb 2009) | 10 lines Merged revisions 174369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines This patch solves some compiler complaints in both 32 and 64-bit environments. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174371 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-09Merged revisions 174327 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 | mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 lines Fix something I messed up in the merge I just did ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174328 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-09Merged revisions 174301 via svnmerge from mmichelson1-2/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb 2009) | 20 lines Merged revisions 174282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174322 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-09Merged revisions 174219 via svnmerge from file1-9/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon, 09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off. (closes issue #14407) Reported by: mostyn ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174220 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-07Merged revisions 174149 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009) | 10 lines Merged revisions 174148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines Fix a race condition that could cause a crash. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174151 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Merged revisions 174084 via svnmerge from dhubbard1-7/+32
https://origsvn.digium.com/svn/asterisk/trunk ................ r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) | 13 lines Merged revisions 174082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa ........ ................ ------------------------------------------------------------------------ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174085 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Blocked revisions 174046 via svnmergedvossel0-0/+0
........ r174046 | dvossel | 2009-02-06 14:12:33 -0600 (Fri, 06 Feb 2009) | 12 lines Adds immediate yes/no option to iax.conf This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well. (closes issue #14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk uploaded by jcovert (license 551) iax.conf.sample.patch uploaded by jcovert (license 551) Tested by: jcovert, dvossel Review: http://reviewboard.digium.com/r/143/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174075 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Merged revisions 174041 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 lines Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription. (closes issue #14322) Reported by: amessina ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174042 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Merged revisions 173974 via svnmerge from file1-42/+39
https://origsvn.digium.com/svn/asterisk/trunk ................ r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) | 15 lines Merged revisions 173967-173968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger ........ r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a debug message I put in by accident. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173986 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Merged revisions 173952 via svnmerge from mnicholson1-1/+62
https://origsvn.digium.com/svn/asterisk/trunk ................ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb 2009) | 14 lines Merged revisions 173917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173963 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Blocked revisions 173902 via svnmergefile0-0/+0
........ r173902 | file | 2009-02-06 11:59:17 -0400 (Fri, 06 Feb 2009) | 4 lines Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached. (closes issue #14414) Reported by: bluecrow76 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173903 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Blocked revisions 173858 via svnmergerussell0-0/+0
........ r173858 | russell | 2009-02-06 04:55:35 -0600 (Fri, 06 Feb 2009) | 13 lines Add a common implementation of a scheduler context with a dedicated thread. This commit expands the Asterisk scheduler API to include a common implementation of a scheduler context being processed by a dedicated thread. chan_iax2 has been updated to use this new code. Also, as a result, this resolves some race conditions related to the previous chan_iax2 scheduler handling. Related to rev 171452 which resolved the same issues in 1.4. Code from team/russell/sched_thread2 Review: http://reviewboard.digium.com/r/129/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173859 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Blocked revisions 173848 via svnmergerussell0-0/+0
........ r173848 | russell | 2009-02-06 04:25:09 -0600 (Fri, 06 Feb 2009) | 2 lines Resolve a memory leak that would occur on an invalid channel given to Action: Status ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173849 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173776 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu, 05 Feb 2009) | 14 lines Update extensions.conf.sample to be correct. In trunk, the only necessary change pointed out was that the call to ChanIsAvail uses an option that has been removed. For the 1.6.1 branch, however, it appears that the sample file is badly in need of updating since there are |'s used all over the place there. My tentative plan is just to copy trunk's sample config file to those branches since the info there is most up-to-date and should be correct for use in 1.6.1 Thanks to macli in #asterisk-dev for bringing this up ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173777 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173773 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb 2009) | 7 lines Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage (closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173774 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173697 via svnmerge from jpeeler1-2/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600 (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173698 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173693 via svnmerge from mmichelson1-6/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb 2009) | 20 lines Merged revisions 173692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173694 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173593 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600 (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173594 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173589 via svnmerge from mmichelson1-5/+88
https://origsvn.digium.com/svn/asterisk/trunk ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600 (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173590 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05fix WORKING_FORK detectionjpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173554 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05regenerate with bootstrap.shtilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173548 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04I messed up and accidentally reverted the trunk-merged prop before ↵jpeeler0-0/+0
committing 173546. Added it manually. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173547 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173500 via svnmerge from jpeeler1-38/+65
https://origsvn.digium.com/svn/asterisk/trunk ................ r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) | 23 lines Merged revisions 173211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173546 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173507 via svnmerge from mmichelson1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 | mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 lines Fix some areas where the incorrect interface was passed to ast_device_state I swear it feels like I already did this once... (closes issue #14359) Reported by: francesco_r ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173534 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173458 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 | tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines When using a socket as a FILE *, the stdio functions will sometimes try to do an fseek() on the stream, which is an invalid operation for a socket. Turning off buffering explicitly lets the stdio functions know they cannot do this, thus avoiding a potential error. (closes issue #14400) Reported by: fnordian Patches: tcptls.patch uploaded by fnordian (license 110) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173460 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173397 via svnmerge from mmichelson1-3/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb 2009) | 11 lines Merged revisions 173396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines Revert my previous change because it was stupid ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173398 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173393 via svnmerge from mmichelson1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb 2009) | 11 lines Merged revisions 173392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever matter, but it's needed. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173394 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173354 via svnmerge from mmichelson1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 | mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30 lines Fix a problem where file playback would cause fds to remain open forever The problem came from the fact that a frame read from a format interpreter was not freed. Adding a call to ast_frfree fixed this. The explanation for why this caused the problem is a bit complex, but here goes: There was a problem in all versions of Asterisk where the embedded frame of a filestream structure was referenced after the filestream was freed. This was fixed by adding reference counting to the filestream structure. The refcount would increase every time that a filestream's frame pointer was pointing to an actual frame of data. When the frame was freed, the refcount would decrease. Once the refcount reached 0, the filestream was freed, and as part of the operation, the open files were closed as well. Thus it becomes more clear why a missing ast_frfree would cause a reference leak and cause the files to not be closed. You may ask then if there was a frame leak before this patch. The answer to that is actually no! The filestream code was "smart" enough to know that since the frame we received came from a format interpreter, the frame had no malloced data and thus didn't need to be freed. Now, however, there is cleanup that needs to be done when we finish with the frame, so we do need to call ast_frfree on the frame to be sure that the refcount for the filestream is decremented appropriately. (closes issue #14384) Reported by: fiddur Patches: 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, putnopvut ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173355 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173311 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 | tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10 lines Ensure that commas placed in the middle of extension character classes do not interfere with correct parsing of the extension. Also, if an unterminated character class DOES make its way into the pbx core (through some other method), ensure that it does not crash Asterisk. (closes issue #14362) Reported by: Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173312 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-03Fixes issue with IAX2 transfer not handing of calls. dvossel1-8/+74
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. (issue #13468) Review: http://reviewboard.digium.com/r/140/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173250 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-03Merged revisions 173104 via svnmerge from tilghman1-1/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600 (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02Merged revisions 173067 via svnmerge from twilson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) | 9 lines Merged revisions 173066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines Fix a feature inheritance bug I added after code review ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173068 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02Merged revisions 172894 via svnmerge from lmadsen1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02 Feb 2009) | 7 lines Update the res_ldap.conf file with a better working example. (closes issue #13861) Reported by: scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10) Tested by: jcovert ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172896 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02Blocked revisions 172890 via svnmergemurf0-0/+0
........ r172890 | murf | 2009-02-02 10:37:15 -0700 (Mon, 02 Feb 2009) | 41 lines This change allows the disconnect feature (as in "one-touch" in features.c) to be used within the dial app, before a call is bridged. Many thanks to sobomax for submitting this patch. Quoting from bug 11582: "So the goal of the patch was to use the user configured feature code during the call setup phase. The original ast_feature_interpret() function is not well suited for this purpose as it uses much call bridge specific data and doesn't separate a detection of feature from a feature handler call. So a new function ast_feature_detect() has been extracted off the ast_feature_interpret() function but keeping the original logic intact except some insignificant changes to locking. "Having created the ast_feature_detect() function the possibility to use feature detection in almost any place of the asterisk code. So a call to this function has been added to wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler however and uses old call leg disconnect logic to make the changes as small and simple as possible to prevent unexpected problems. A disconnect feature currently is the only one supported during call setup as other features as call parking and call transfer don't make much sense during call setup. However if need in some of the features would arise it is much easier to implement as the infrastructure changes are already in place with this patch." I have cleaned up the patch somewhat, and verified that the existing functionality is not harmed, and that the new functionality works. Terry has committed his stuff, and there were no conflicts (see 14274). (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) patch-include__asterisk__features.h uploaded by sobomax (license 359) patch-res__res_features.c uploaded by sobomax (license 359) enable-features-during-call-setup.diff uploaded by sobomax (license 359) 11583.newdiff uploaded by murf (license 17) enable-features-during-call-setup-1.diff uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) Tested by: sobomax, murf ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172892 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-01Merged revisions 172741 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) | 4 lines Blank argument crashes Asterisk (closes issue #14377) Reported by: amorsen ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172742 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-31Merged revisions 172706 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009) | 7 lines Don't increment the loop, now that incrementing is taken care of by the decoder function. (closes issue #14363) Reported by: andrew53 Patches: func_strings_filter.patch uploaded by andrew53 (license 519) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172707 f38db490-d61c-443f-a65b-d21fe96a405b
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