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${ noResults }
16941 Commits (cd84ad1fc07342b6df1d0704fa9335bed0729e79)
Author | SHA1 | Message | Date |
---|---|---|---|
|
cd84ad1fc0 |
Merged revisions 185261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 | russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines Don't free() an astobj2 object. (closes issue #14672) Reported by: makoto ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
d1709254d4 |
Merged revisions 185197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) | 15 lines Merged revisions 185196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines Fix crash when moving audiohooks between channels. Handle the scenario where we are called to move audiohooks between channels and the source channel does not actually have any on it. (closes issue #14734) Reported by: corruptor ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185198 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ae65af4244 |
Merged revisions 185123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185125 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
29aa5c71a5 |
Merged revisions 185122 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines Make chan_misdn BRI TE side normally defer channel selection to the NT side. Channel allocation collisions are not handled by chan_misdn very well. This patch simply avoids the problem for BRI only. For PRI, allocation collisions are still possible but less likely since there are simply more channels available and each end could use a different allocation strategy. misdn.conf options available: te_choose_channel - Use to force the TE side to allocate channels. method - Specify the channel allocation strategy. (closes issue #13488) Reported by: Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes, festr ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185124 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 185072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar 2009) | 45 lines Merged revisions 185031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) | 39 lines Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked. (This is copied and pasted from the review request I made for this patch) Asterisk has some odd behavior when queue weights are used. The current logic used when potentially calling a queue member is: If the member we are going to call is part of another queue and _that other queue has any callers in it_ and has a higher weight than the queue we are calling from, then don't try to contact that member. The issue here is what I have marked with underscores. If the higher-weighted queue has any callers in it at all, then the queue member will be unreachable from the lower-weighted queue. This has the potential to be really really bad if using a queue strategy, such as leastrecent or fewestcalls, with the potential to call the same member repeatedly. The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works well for this situation. With this set of changes, the logic used becomes: If the member we are going to call is part of another queue, the other queue has a higher weight than the queue we are calling from, and the higher weight queue has at least as many callers as available members, then do not try to contact the queue member. If the higher weighted queue has fewer callers than available members, then there is no reason to deny the call to this member since the other queue can afford to spare a member. Since the fix involved writing a generic function for determining the number of available members in the queue, I also modified the is_our_turn function to make use of the new num_available_members function to determine if it is our turn to try calling a member. There is one small behavior change. Before writing this patch, if you had autofill disabled, then if you were the head caller in a queue, you would automatically be told that it was your turn to try calling a member. This did not take into account whether there were actually any queue members available to take the call. Now we actually make sure there is at least one member available to take the call if autofill is disabled. (closes issue #13220) Reported by: garychen Review: http://reviewboard.digium.com/r/202/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185087 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 184986 via svnmerge
................ r184986 | mmichelson | 2009-03-30 10:25:04 -0500 (Mon, 30 Mar 2009) | 27 lines Blocked revisions 184980 via svnmerge ........ r184980 | mmichelson | 2009-03-30 10:23:59 -0500 (Mon, 30 Mar 2009) | 22 lines Backport state interface changes to app_queue from trunk. After several issues raised on the Asterisk bugtracker against the 1.4 branch were determined to be fixable with the state interface change available in the 1.6.X series, it finally came time to just suck it up and backport the change. For a detailed explanation of what this change entails, the original trunk commit for this feature may be found here: http://svn.digium.com/view/asterisk?view=revision&revision=97203 In addition, the details for the use of this change to fix the problems stated in issue #12970 may be found in the review request I made for this change. It is linked below. (closes issue #12970) Reported by: edugs15 Review: http://reviewboard.digium.com/r/116 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184989 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines Merged revisions 184947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines Improve our handling of T38 in the initial INVITE from a device. We now answer with matching media streams to what is requested. If an INVITE is received with both a T38 and RTP media stream this means we answer with both. For any outgoing calls created as a result of this inbound one no T38 is requested in the initial INVITE. Instead if we start receiving udptl packets we trigger a reinvite on the outbound side. (closes issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu Review: http://reviewboard.digium.com/r/208/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184949 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184910 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 Mar 2009) | 4 lines Fix build error when chan_h323 is not being built. (reported by cai1982 in #asterisk-dev) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184911 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009) | 13 lines Merged revisions 184842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) | 5 lines Ensure targs variable is fully initialized. (closes issue #14758) Reported by: tim_ringenbach ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 | russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines Simplify chan_h323 build to not require a second run of "make". (closes issue #14715) Reported by: jthurman Patches: h323-makefile-1.6.0.7-rc2.patch uploaded by jthurman (license 614) h323-makefile-1.6.1.0-rc3.patch uploaded by jthurman (license 614) h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614) Tested by: tzafrir, russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 184762 via svnmerge
........ r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 lines Improve timing interface to remember which provider provided a timer The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error. This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider. (closes issue #14697) Reported by: moy Review: http://reviewboard.digium.com/r/211/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184763 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27 Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure we use the best RNG available. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184727 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184673 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 | file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix speech structure leak in the AGI speech recognition integration. The AGI dialplan applications did not destroy the speech structure automatically if it was not destroyed by the running AGI script. They will now do this. (issue LUMENVOX-15) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184674 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | 16 lines Merged revisions 184565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls. If calls were placed using an IP address or hostname the global nat setting was copied over but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP actions. (closes issue #14546) Reported by: acunningham ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184515 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 | russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines Don't act surprised if we get a -1 indication. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184516 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar 2009) | 9 lines Merged revisions 184447 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar 2009) | 3 lines use new, improved 8kHz prompts ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184389 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r184389 | dvossel | 2009-03-26 16:09:37 -0500 (Thu, 26 Mar 2009) | 14 lines Merged revisions 184388 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF 8 app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent. During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up. (closes issue #12442) Reported by: tzafrir ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184391 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 | file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix issue with a T38 reinvite being sent even if not configured to do so. If we receive a T38 request negotiate control frame we should only attempt to do so if the option is enabled on the dialog. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184281 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) | 19 lines Merged revisions 184188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | 13 lines Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. When moving the cursor backward and pressing TAB to autocomplete, a NULL is put in the line and we are loosing what we have already wrote after the actual cursor position. (closes issue #14373) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) Tested by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184222 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 | russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines Fix build issues on Mac OSX. (closes issue #14714) Reported by: ygor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184148 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar 2009) | 15 lines Merged revisions 184078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. The 'digit' variable is guaranteed to be non-NULL, so the if statement could never evaluate true. Changing to ast_strlen_zero makes the logic correct. This was found while reviewing ast_channel_ao2 code review. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184080 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low and =medium The default codec configuration for chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as the codec in some test calls, but that no longer happens after this change. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500 (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183915 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183865 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 | tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines Allow browsers to cache images and other static content. (This is a regression over 1.4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar 2009) | 13 lines Merged revisions 183700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines Fix a memory leak in res_monitor.c The only way that this leak would occur is if Monitor were started using the Manager interface and no File: header were given. Discovered while reviewing the ast_channel_ao2 review request. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183767 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009) | 7 lines Fixes a documentation error introduced during the CLI cleanup at AstriDevCon 2008. (closes issue #14655) Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183702 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183560 | russell | 2009-03-20 12:00:58 -0500 (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) | 2 lines Fix a crash in IAX2 registration handling found during load testing with dvossel. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 183553-183555 via svnmerge
........ r183553 | mmichelson | 2009-03-20 11:19:53 -0500 (Fri, 20 Mar 2009) | 5 lines Add some missing symbols to main/asterisk.exports Hey! chan_sip.so loads now! ........ r183554 | mmichelson | 2009-03-20 11:24:20 -0500 (Fri, 20 Mar 2009) | 3 lines Remove symbols I just added to main/asterisk.exports and instead rename the functions. ........ r183555 | mmichelson | 2009-03-20 11:25:17 -0500 (Fri, 20 Mar 2009) | 3 lines Fix chan_sip so it builds. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183556 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 183511 via svnmerge
........ r183511 | eliel | 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) | 2 lines Remove duplicate <description> inside the xml documentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183532 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines Merged revisions 183386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183437 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500 (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines Delay signalling progress until a PRI channel really signals progress. (closes issue #13034) Reported by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by tilghman (license 14) patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183244 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 | mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 lines Fix a memory leak associated with queues. For every attempt that app_queue made to place an outbound call to a queue member, we would allocate a queue_end_bridge structure. When the bridge for the call had completed, we would free the structure. Unfortunately not all call attempts actually end up bridged to a member, so we need to be more selective of when to allocate the structure. With this change, the allocation occurs in an area where we can guarantee that the call will be bridged. (closes issue #14680) Reported by: caspy Patches: 14680.patch uploaded by mmichelson (license 60) Tested by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183245 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) | 10 lines Merged revisions 183241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving like expected. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183243 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 183117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use." A user was having an issue where if an outgoing SIP call was canceled, the SIP device would remain in use if we had not received any response to the initial INVITE we sent out. The SIP device would remain in use until the autocongestion timer was exhausted. I tracked down the cause of this to be the section of code I am removing here. I asked several people what the purpose of this code was meant to be, but no one could give me any sort of answer as to why this was here. The person who was having this issue has been using this patch for several months and it has stopped the problems they have had. AST-196 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 183108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines Improve our triggering of a T38 switchover internally when triggered by a received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 183057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 | file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix an issue where a T38 control frame would get dropped. If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183066 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 183028 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183029 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 182847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 182722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182723 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 | russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines Tweak the handling of the frame list inside of ast_answer(). This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 | kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 lines correct logic flaw in ast_answer() changes in r182525 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182532 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182525 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182526 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Blocked revisions 182521 via svnmerge
........ r182521 | seanbright | 2009-03-17 10:24:53 -0400 (Tue, 17 Mar 2009) | 3 lines Don't include a space before the optional extra text that may follow a help string. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) | 14 lines Merged revisions 182449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182451 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Blocked revisions 182408 via svnmerge
........ r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009) | 8 lines OPENR2 uses an incorrect string value if the extension delimiter is not present. * Fixed OPENR2 using an incorrect string value if the extension delimiter is not present in the Dial() function. This was fixed for SS7 and PRI in trunk -r172400. * Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others. * Removed trailing whitespace that appeared with OPENR2. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Blocked revisions 182362 via svnmerge
........ r182362 | russell | 2009-03-16 15:53:21 -0500 (Mon, 16 Mar 2009) | 2 lines Update UPGRADE.txt and CHANGES for 1.6.3 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 182355 via svnmerge
........ r182355 | russell | 2009-03-16 15:35:58 -0500 (Mon, 16 Mar 2009) | 29 lines Add MFC/R2 support for chan_dahdi. This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182356 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500 (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182283 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182278 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182278 | tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution. Previously, FILE() returned one less character than specified, due to the terminating NULL. Both the offset and length parameters now behave identically to the way variable substitution offsets and lengths also work. (closes issue #14670) Reported by: BMC ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182279 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |