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2009-08-18git-svn-id: http://svn.digium.com/svn/asterisk/tags/ ↵v1.6.0.6kpfleming6-12/+12
2009-02-23Update the .version and ChangeLog files for the release.lmadsen2-1/+5
git-svn-id: http://svn.digium.com/svn/asterisk/tags/ f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-23Create the tag from the release candidatelmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/ f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Use autotagged externalslmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/ f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Importing files for releaselmadsen3-0/+49430
git-svn-id: http://svn.digium.com/svn/asterisk/tags/ f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Creating tag for the release of asterisk-
git-svn-id: http://svn.digium.com/svn/asterisk/tags/ f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175591 via svnmergemmichelson0-0/+0
................ r175591 | mmichelson | 2009-02-13 13:49:38 -0600 (Fri, 13 Feb 2009) | 22 lines Merged revisions 175590 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines Fix a potential crash situation when using IMAP voicemail If calling into VoiceMailMain when using IMAP storage, it was possible to crash Asterisk by hanging up the phone when prompted for a voicemail mailbox. This patch fixes the issue. While it may appear that this patch is superficial, it allows code execution to continue to the failure case just below the IMAP_STORAGE code block where this patch has been applied (closes issue #14473) Reported by: dwpaul Patches: voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175592 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Merged revisions 175549 via svnmerge from file1-3/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 | file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add an option to keep the recorded file upon hangup. (closes issue #14341) Reported by: fnordian ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175550 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175512 via svnmergekpfleming0-0/+0
........ r175512 | kpfleming | 2009-02-13 07:41:52 -0600 (Fri, 13 Feb 2009) | 3 lines document G.722.1/.1C support ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175514 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175508 via svnmergekpfleming0-0/+0
........ r175508 | kpfleming | 2009-02-13 07:35:24 -0600 (Fri, 13 Feb 2009) | 15 lines Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14) This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175510 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175475 via svnmergedhubbard0-0/+0
........ r175475 | dhubbard | 2009-02-12 22:22:35 -0600 (Thu, 12 Feb 2009) | 1 line add 'faxbuffers' configuration option information to CHANGES ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175476 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Blocked revisions 175411 via svnmergedhubbard0-0/+0
........ r175411 | dhubbard | 2009-02-12 18:13:38 -0600 (Thu, 12 Feb 2009) | 13 lines Add dynamic fax buffer configuration option to chan_dahdi.conf When the 'faxdetect' configuration option is used, one may also want to use the 'faxbuffers' configuration option in chan_dahdi.conf. This option will dynamically use the configured 'faxbuffers' buffer policy on a channel for the life of the call following the detection of fax tones. The faxbuffers buffer policy will be reverted during call teardown. An example use of 'faxbuffers' is below. This example would switch to using 6 buffers with a full buffer policy. faxbuffers=>6,full ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175472 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175368 via svnmerge from russell1-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines Remove useless string copy, and make sscanf safe again ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175369 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Blocked revisions 175344 via svnmergedvossel0-0/+0
........ r175344 | dvossel | 2009-02-12 15:27:11 -0600 (Thu, 12 Feb 2009) | 10 lines Adds force encryption option to iax.conf This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well. (closes issue #13285) Reported by: sgofferj Tested by: russell Review: http://reviewboard.digium.com/r/150/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175366 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175334 via svnmerge from tilghman1-23/+51
https://origsvn.digium.com/svn/asterisk/trunk ................ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) | 16 lines Merged revisions 175311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175347 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Fix mistake in merging conflict from 175299.jpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175301 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175298 via svnmerge from jpeeler1-1/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175299 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175295 via svnmerge from russell1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines Avoid using ast_strdupa() in a loop. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175296 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175255 via svnmerge from russell1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) | 4 lines Don't enable something by default that has a dependency on something _not_ enabled by default. menuselect was not happy with this. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175256 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Blocked revisions 175250 via svnmergekpfleming0-0/+0
........ r175250 | kpfleming | 2009-02-12 12:48:52 -0600 (Thu, 12 Feb 2009) | 1 line correct warning message to not refer specifically to DAHDI ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175252 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175188 via svnmerge from jpeeler1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175189 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175125 via svnmerge from russell1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines Merged revisions 175124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175126 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175121 via svnmerge from mmichelson2-2/+23
https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 | mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 lines Make lock information for ao2_trylock be more useful and gnarly Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175122 f38db490-d61c-443f-a65b-d21fe96a405b
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