mirror of https://github.com/asterisk/asterisk
master
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${ noResults }
13469 Commits (efb22ba096ddfc0803f0275d2ccbb22a62d85566)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
efb22ba096 |
Add a similar dependency on SMDI for voicemail as already exists for ADSI.
(closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
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ee5ca5fb33 |
Fix a typo which caused loss of audio when using G729 in some scenarios with a smoother present.
(closes issue #15105) Reported by: bamby Patches: process-vad-correctly.diff uploaded by bamby (license 430) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195206 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ac71a26c0f |
Fix a bug where the codecs of the called party leg were not properly sent back to the caller call leg when reinvited.
(closes issue #13569) Reported by: bkw918 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195095 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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85483848d0 |
Don't try to unlock a bogus channel.
(closes issue #15144) Reported by: cristiandimache git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195020 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ca3481edb9 |
IAX2 REGAUTH loop
IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001) (closes issue #14867) Reported by: aragon Tested by: dvossel (closes issue #14717) Reported by: mobeck Patches: regauth_loop_update_patch.diff uploaded by dvossel (license 671) Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5b9004d067 |
Fix some spelling fail.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194764 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1e410cdfc5 |
Update to previous IAX2 "Ghost" Channels patch.
Fixed some comments made on reviewboard for the previous patch. (issue #14207) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194685 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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616674ae68 |
IAX2 "Ghost" Channels
There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away. (closes issue #14207) Reported by: clive18 Review: https://reviewboard.asterisk.org/r/246/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194557 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9a14ec91a7 |
Update URL to Reviewboard
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7aa29c797a |
Fix a race condition where a reinvite could trigger a 482 response.
The loop detection/spiral detection code in chan_sip used the owner channel's state as a criterion for determining if the incoming INVITE is a looped request. The problem with this is that the INVITE-handling code happens in a different thread than the thread that marks the owner channel as being up. As a result, if a reinvite were to come in very quickly, say from another Asterisk on the same LAN, it was possible for the reinvite to arrive before the owner channel had been set to the up state. This patch corrects the problem by using the invitestate of the sip_pvt instead, since that can be guaranteed to be set correctly by the time the reinvite arrives. Since there is a switch statement further in the INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate of the sip_pvt in case we should actually be treating the channel as if it were up already. (closes issue #12215) Reported by: jpyle Patches: 12215_confirmed.patch uploaded by mmichelson (license 60) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194484 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7189d75805 |
Remove an extraneous unlocking operation from ast_channel_free.
In the case that we could not remove the desired channel from the list of channels, there was an extra call to unlock the channel list. Since we unlock the list later on in the function anyway, this results in the list being unlocked twice yet only being locked once. (closes issue #15098) Reported by: tim_ringenbach Patches: remove_extra_unlock.diff uploaded by tim (license 540) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194356 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0e29c52c46 |
Pull in a piece of murf's 88166 patch that makes it safe to call
pbx_substitute_variables_helper_full with a non-zero'd buffer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194322 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0a55b56dc8 |
Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over.
(closes issue #14815) Reported by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue #14460) Reported by: moliveras Tested by: moliveras git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194208 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c23d9c8960 |
Fix logic for how to proceed with a single digit extension.
(closes issue #15091) Reported by: andrew Patches: 20090512__issue15091.diff.txt uploaded by tilghman (license 14) Tested by: andrew git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194137 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bec8573c37 |
This change modifies app_queue to properly generate CDR records in failure
situations. This involves setting a proper cdr disposition coresponding to the given failure condition and ensuring the proper information is stored in the cdr record. (closes issue #13691) Reported by: dferrer Tested by: mnicholson (closes issue #13637) Reported by: atis Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f6ba2472bd |
Avoid initializing routines if the authentication fails. Fixes a crash (RR) issue.
(closes issue #14508) Reported by: tiziano Patches: 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license 377) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193955 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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63c0dca7bd |
Set the invitestate to INV_CANCELLED only if we are actually sending a SIP CANCEL.
The problem was that the hangup code was setting the invitestate too early. The result of this was that we would always send a CANCEL request, even if it was not an appropriate time to do so (e.g. we have not yet received a provisional response for our INVITE). Note that this same fix had been applied to trunk and the 1.6.X branches starting with revision 155467. This is why you will see this revision being blocked from those places. AST-216 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193880 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8425d87bdf |
Move 300 bytes around on the stack, to make more room for an extension buffer.
This allows more concurrent extensions to be copied for a single voicemail, without creating a possibility of upsetting existing users, where a dialplan could run out of stack space where it had run fine before. Alternatively, we could have allocated off the heap, but that is a larger change and would have increased the chance for instability introduced by this change. This is really solved starting in 1.6.0.11, as the use of an ast_str buffer allows an unlimited number of extensions (up to available memory). We additionally create a new warning message when the buffer length is exceeded, permitting administrators to see an issue after the fact, whereas previously the list was silently truncated. (closes issue #14739) Reported by: p_lindheimer Patches: 20090417__bug14739.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193755 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4d64b0c937 |
Sent wrong message to clear a call we started if the other end has not responed yet.
In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet),
it is not allowed to clear the call with RELEASE_COMPLETE. It must be
cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer
to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b)
Patches:
chan-misdn-ccstate7.patch uploaded by customer.
JIRA ABE-1862
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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17 years ago |
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b0c91953cf |
Document CHANNEL(transfercapability) in CLI documentation.
(issue #15073) Reported by: pkempgen Patches: 20090511__issue15073.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193544 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4c8e4a2969 |
Set the proper disposition on originated calls.
(closes issue #14167) Reported by: jpt Patches: call-file-missing-cdr2.diff uploaded by mnicholson (license 96) Tested by: dlotina, rmartinez, mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193391 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3d0faa34ca |
"misdn show config" segfaults asterisk, if no MSN lists
(closes issue #14976) Reported by: alecdavis Patches: misdn_config.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, FabienToune git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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58b5a85e80 |
Make absolute paths for logger channels work properly
(Note: This is not a new feature, it was previously undocumented and broken.) The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193193 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6a5cbae87f |
Fix Background within a Macro for FreePBX.
If the single digit DTMF is an extension in the specified context, then go there and signal no DTMF. Otherwise, we should exit with that DTMF. If we're in Macro, we'll exit and seek that DTMF as the beginning of an extension in the Macro's calling context. If we're not in Macro, then we'll simply seek that extension in the calling context. Previously, someone complained about the behavior as it related to the interior of a Gosub routine, and the fix (#14011) inadvertently broke FreePBX (#14940). This change should fix both of these situations, but with the possible incompatibility that if a single digit extension does not exist (but a longer extension COULD have matched), it would have previously gone immediately to the "i" extension, but will now need to wait for a timeout. (closes issue #14940) Reported by: p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0971bac5bc |
Give a more helpful message when an incoming call's dialed extension does not match.
Added the dialed extension and context to the chan_misdn messages warning that the dialed number cannot be matched in the dialplan. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193050 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c57efbe571 |
Eliminate repetition of fullcontact during reconstruction.
If the fullcontact field appears in both the sippeers and the sipregs table, then during reconstruction of the field, it will otherwise be doubled. (closes issue #14754) Reported by: Alexei Gradinari Patches: 20090506__bug14754.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192932 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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84f708474a |
Make ParkedCall application stop execution of the dialplan after hang up
Just changed park_exec to always return non-zero. I really wasn't entirely sure at first if this was a bug. Decided it was since it would be surprising when not using ParkedCall in the dialplan to hang up and have dialplan execution continue. (closes issue #14555) Reported by: francesco_r git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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202bc9464e |
Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled.
(closes issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192633 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c489a2f6a3 |
Fix Javascript error when using astman.js in Internet Explorer.
Internet Explorer (tested with 7.0) does not like trailing commas on constructs like object initializers, so get rid of them to avoid some errors. (closes issue #15026) Reported by: rajnishgiri Patches: bug15026.patch uploaded by seanbright (license 71) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192524 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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385e28f532 |
Fix an incorrect assumption that certain values on the channel will always exist when they may not.
The CDR code involved with bridges wrongly assumed that the currently executing application and data values will always exist. It is possible for this to be false when call forwarding is involved. (closes issue #14984) Reported by: gincantalupo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192454 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6b15b32783 |
Fix a bug where the followme application would continue trying numbers after the caller hung up.
(closes issue #13624) Reported by: sgenyuk git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192429 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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dcb712422a |
global mohinterpret setting is ignored
mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers. (closes issue #14728) Reported by: dimas Patches: v1-14728.patch uploaded by dimas (license 88) Tested by: dimas, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192213 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b67282e2fd |
Fix a bug which resulted from the Hebrew voicemail commit.
This fixes a case where a certain message could get played twice. (closes issue #13155) Reported by: greenfieldtech Patches: app_voicemail.c.multi-lang-patch uploaded by greenfieldtech (license 369) Tested by: greenfieldtech git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191778 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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972d9bf53c |
Kevin has informed me that thi sort of thing is not necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191629 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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85a8916552 |
Move static buffers to outside for loops in app_chanspy.
Similar to seanbright's commit 191422, this moves some static buffers to be defined outside of for loops since it is undefined if memory will be re-used or if the stack will grow with each iteration of the loop. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191628 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c2d8897257 |
SIP Response 410 maps to cause code 22 (or 23), not 1.
(closes issue #14993) Reported by: BigJimmy Patches: causepatch uploaded by BigJimmy (license 371) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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98c23ee3cf |
Fix DTMF not being sent to other side after a partial feature match
This fixes a regression from commit 176701. The issue was that ast_generic_bridge never exited after the feature digit timeout had elapsed, which prevented the queued DTMF from being sent to the other side. This issue was reported to me directly. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191488 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b5ec450104 |
Move the defintion of the a couple arrays out of loops.
According to Kevin, it is unspecified as to whether a variable defined inside a block is allocated once by the compiler or for each pass through the block (loops being the only interesting case), so just define these before we get into our loop to be sure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191422 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f8b1da1872 |
Allow H.323 to compile with FDLEAK checking enabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4a734fd926 |
Patch to fix tab-completion crash on "remove extension"
This patch simply removes some old code back before Asterisk used editline. This fixes the crash that occurred when tab-completing "remove extension". (closes issue #14689) Reported by: isaacgal git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191096 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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603a56aa69 |
Fix a crash in app_queue with very long member lists.
A user reported via #asterisk that with very long lists of members, a crash occurs in ast_strdupa, so just use a single buffer and ast_copy_string instead of stack allocating copys of each interface name. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7c82c2b240 |
Fix 'inconsistent line endings' when autoconf 2.63 is used
Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@190721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1c1d4a80e0 |
Fix a typo from 190661.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@190662 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8798a0f9dd |
Resolve a crash in res_smdi when used with chan_dahdi.
When chan_dahdi goes to get an SMDI message, it provides no search criteria. It just grabs the next message that arrives. This code was written with the SMDI dialplan functions in mind, since that is now the preferred method of using SMDI. However, this broke support of it being used from chan_dahdi. (closes AST-212) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@190661 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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03eb22fe76 |
Remove a bogus ast_channel_unlock().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@190356 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a8a55273cf |
Fix a bug in chan_local glare hangup detection.
If both sides of a Local channel were hung up at around the same time it was possible for one thread to destroy the local private structure and have the other thread immediately try to remove the already freed structure from the local channel list. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@190286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1dac2a69e2 |
unistd.h is required for usleep() on Darwin. It will not hurt to include it always
on other platforms either. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@190187 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c922eca9a8 |
Detect availability of pthread_rwlock_timedwrlock() before using it.
(closes issue #14930) Reported by: tilghman Patches: 20090420__bug14930.diff.txt uploaded by tilghman (license 14) Tested by: mvanbaak, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@190092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a1b5f4a67d |
Make chan_h323 respect packetization settings
Previously, packetization settings were ignored and now they are not. A new config option 'autoframing' has been added to mirror the way chan_sip handles it. Turning on the autoframing option (available both as a global option or per peer) overrides the local settings with the remote packetization settings. Testing was performed with varying packetization levels with the following codecs: ulaw, alaw, gsm, and g729. (closes issue #12415) Reported by: pj Patches: 2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7), modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4758539ec5 |
replace sed with tr to remove \r from downloaded file
On some systems, sed does not recognize \r in the pattern the way it was used here. Use tr instead because this works the same across systems. (closes issue #14936) Reported by: leobrown Patches: 2009042201_14936.diff.txt uploaded by mvanbaak (license 7) Tested by: leobrown, mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189849 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |