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${ noResults }
17416 Commits (0a792331bff9bf6568d485030c7a214e20285d2c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
0a792331bf |
Merged revisions 175597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176248 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
e32921415f |
Assist proper thread synchronization when stopping the logger thread.
I was finding that on my dev box, occasionally attempting to "stop now" in trunk would cause Asterisk to hang. I traced this to the fact that the logger thread was waiting on a condition which had already been signalled. The logger thread also need to be sure to check the value of the close_logger_thread variable. The close_logger_thread variable is only checked when the list of logmessages is empty. This allows for the logger thread to print and free any pending messages before exiting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176174 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
6cc10eb351 |
Can't set debug level 2 (intense debugging) unless the syntax matches
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176138 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
0aa256715c |
Remove chan_features.
Review: http://reviewboard.digium.com/r/161/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176100 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
8d00e7a6ed |
Merged revisions 176029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176030 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
96326f5aa1 |
Make the causes array static, and remove the type name as it is not needed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175983 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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115c6abef4 |
Merged revisions 175921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
8c75380f52 |
Make ast_sched_report() and ast_sched_dump() thread safe.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175882 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ca9d3b8ac9 |
Fix a number of problems with ast_sched_report().
1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175829 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cbe45a6e06 |
Merged revisions 175825 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb 2009) | 2 lines format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175827 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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798e83aecb |
Merged revisions 175792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb 2009) | 2 lines Disable format_ilbc.so by default, like codec_ilbc.so ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175801 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c9a8142e58 |
Merged revisions 175777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2 lines Make sure that the debug line is not printed on debug level 0 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175783 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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4ab40e9a10 |
Blocked revisions 175698 via svnmerge
........ r175698 | qwell | 2009-02-13 15:53:16 -0600 (Fri, 13 Feb 2009) | 1 line Zaptel is not DAHDI. Rather, Zaptel is actually Zaptel. (in case you're confused, DAHDI is still DAHDI) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3c9667ae12 |
Merge queue-reset branch to Asterisk
From a user point-of-view, this adds new CLI commands and Manager Actions to better facilitate the reloading of queues and the resetting of their statistics. The new CLI commands are the "queue reload" and "queue reset stats" commands. The new manager actions are the QueueReload and QueueReset commands. Review: http://reviewboard.digium.com/r/115 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b5bde28d9f |
Add manager events for chanspy starting or stopping
(closes issue #14469) Reported by: caio1982 Patches: chanspy_events2.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175655 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
2800100cf7 |
fix a few more XML documentation problems
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175636 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6a0773602a |
add missing </para>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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35ac1d7e1c |
Fixed iax2 key rotation backwards compatibility
Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. Review: http://reviewboard.digium.com/r/159/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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10281bf183 |
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: https://origsvn.digium.com/svn/asterisk/trunk@175591 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
0cfdaa2dc0 |
Add an option to keep the recorded file upon hangup.
(closes issue #14341) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175549 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3854faf2d7 |
document G.722.1/.1C support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2a53f2ec98 |
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: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
1981fdac02 |
add 'faxbuffers' configuration option information to CHANGES
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175475 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d11e6f0591 |
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: https://origsvn.digium.com/svn/asterisk/trunk@175411 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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015c7189fb |
Blocked revisions 175407 via svnmerge
........ r175407 | mmichelson | 2009-02-12 17:22:44 -0600 (Thu, 12 Feb 2009) | 12 lines Fix a place where filestreams were not refcounted properly This section was already present in trunk and other branches, but did not exist in 1.4. (closes issue #14395) Reported by: ZX81 Patches: 14395.patch uploaded by putnopvut (license 60) Tested by: ZX81 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175408 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a741658f58 |
Remove useless string copy, and make sscanf safe again
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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178e6f06df |
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: https://origsvn.digium.com/svn/asterisk/trunk@175344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9e0f5d2afe |
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: https://origsvn.digium.com/svn/asterisk/trunk@175334 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a46d290802 |
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: https://origsvn.digium.com/svn/asterisk/trunk@175298 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
768c73160e |
Avoid using ast_strdupa() in a loop.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4b16e0f051 |
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: https://origsvn.digium.com/svn/asterisk/trunk@175255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1448b5db6a |
correct warning message to not refer specifically to DAHDI
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175250 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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66e88633a5 |
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: https://origsvn.digium.com/svn/asterisk/trunk@175188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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53f3ab973e |
Setting key rotation to be off by default
Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175127 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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12f02a8c11 |
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: https://origsvn.digium.com/svn/asterisk/trunk@175125 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3a9d79f056 |
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: https://origsvn.digium.com/svn/asterisk/trunk@175121 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c873d72ca2 |
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: https://origsvn.digium.com/svn/asterisk/trunk@175089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7bb5ef8399 |
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: https://origsvn.digium.com/svn/asterisk/trunk@175058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0d5da5f436 |
Fix a bit of odd logic for announcing position. Sync with 1.6.0's logic
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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34161542e9 |
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: https://origsvn.digium.com/svn/asterisk/trunk@174948 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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47ebea6a8d |
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: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ff0b908777 |
Blocked revisions 174885 via svnmerge
........ r174885 | tilghman | 2009-02-11 14:54:18 -0600 (Wed, 11 Feb 2009) | 13 lines Restore a behavior that was recently changed, when we fixed issue #13962 and issue #13363 (related to issue #6176). When a hangup occurs during a Macro execution in earlier 1.4, the h extension would execute within the Macro context, whereas it was always supposed to execute only within the main context (where Macro was called). So this fix checks for an "h" extension in the deepest macro context where a hangup occurred; if it exists, that "h" extension executes, otherwise the main context "h" is executed. (closes issue #14122) Reported by: wetwired Patches: 20090210__bug14122.diff.txt uploaded by Corydon76 (license 14) Tested by: andrew ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174886 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8435535300 |
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: https://origsvn.digium.com/svn/asterisk/trunk@174844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c9670b24a4 |
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: https://origsvn.digium.com/svn/asterisk/trunk@174805 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e909bfcea9 |
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: https://origsvn.digium.com/svn/asterisk/trunk@174764 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6304c09149 |
Only decrease inringing count if above zero.
(issue #13238) Reported by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174710 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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23939e54f3 |
improve slinfactory API to remove implicit sample rate and require explicit sample rate selection by creator of the slinfactory
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174705 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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656110e42b |
Blocked revisions 174644 via svnmerge
........ r174644 | file | 2009-02-10 14:50:50 -0400 (Tue, 10 Feb 2009) | 6 lines Go off hold when we get an empty reinvite telling us to. (closes issue #14448) Reported by: frawd Patches: hold_invite_nosdp.patch uploaded by frawd (license 610) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174645 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b7dc074376 |
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: https://origsvn.digium.com/svn/asterisk/trunk@174584 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8e6780a5b1 |
Set the type for the peer structure to be a peer as the default.
(closes issue #14447) Reported by: triccyx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |