Commits

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Author Commit Message Labels Comments Date
rmudgett
Fix callerid of Originated calls. Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix ........ Merged revisions 357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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10.3.0-rc3
Branches
10
twilson
Copy CDR variables when set during a bridge This patch makes sure amaflags, accountcode, and userfield get copied to the bridge CDR when set during a bridge (like via a custom feature). (closes issue ASTERISK-16990) Review: https://reviewboard.asterisk.org/r/1721/ ........ Merged revisions 356963 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Branches
10
jrose
Remove possible segfaults from res_odbc by adding locks around usage of odbc handle (closes issue ASTERISK-19011) Reported by: Walter Doekes Patches: issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch uploaded by Walter Doekes (license 5674) review: https://reviewboard.asterisk.org/r/1719/ review: https://reviewboard.asterisk.org/r/1622/ ........ Merged revisions 356917 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Branches
10
mjordan
Fix crash in app_voicemail during close_mailbox In r354890, a memory leak in app_voicemail was fixed by properly disposing of the allocated heard/deleted pointers. However, there are situations, particularly when no messages are found in a folder, where these pointers are not allocated and not NULL. In that case, an invalid free would be attempted, which could crash app_voicemail. As there are a number of code paths where this could occur, t…
Branches
10
rmudgett
Fix worker thread resource leak in SIP TCP/TLS. The SIP TCP/TLS worker threads were created joinable but noone could join them if they died on their own. * Fix the SIP TCP/TLS worker threads to not be created joinable. * _sip_tcp_helper_thread() only needs one parameter since the pvt parameter is only passed in as NULL and never used. (closes issue ASTERISK-19203) Reported by: Steve Davies Review: https://reviewboard.asterisk.org/r/1714/ ..…
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mjordan
Remove srtp_shutdown from res_srtp The patch for ASTERISK-19253 included properly shutting down the libsrtp library in the case of module unload. Unfortunately, not all distributions have the srtp_shutdown call. As such, this patch removes calling srtp_shutdown. ........ Merged revisions 356650 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Branches
10
mjordan
Allow SRTP policies to be reloaded Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but …
Branches
10
rmudgett
Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension. Custom parking extensions may not be coded such that the first and only extension priority is the Park application. These custom parking extensions will not be recognized as parking extensions. When a call is blind transferred to an extension that is not recognized as a parking extension, the normal blind transfer code causes the transferred channel to start executing dialplan. Calls that get p…
Branches
10
mmichelson
Fix ACK routing for non-2xx responses. When we send an ACK for a 2xx response to an INVITE, we are supposed to use the learned route set. However, when we receive a non-2xx final response to an INVITE, we are supposed to send the ACK to the same place we initially sent the INVITE. We had been doing this up until the changes went in that would build a route set from provisional responses. That introduced a regression where we would use…
Branches
10
pabelanger
Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 356430 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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pabelanger
Multiple revisions 356290,356335,356337 ........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed, 22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) Review: https://reviewboard.asterisk.org/r/1763/ ........ r356335 | pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2 lines Add back strsep() function for previous commit ........ r356337 | pabelanger | 2012-02-22 16:36:37 …
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twilson
Track module use count for res_calendar If the res_calendar module was followed immediately by one of the calendar tech modules and "core stop gracefully" was run, Asterisk would crash. This patch adds use count tracking for res_calendar so that it is unloaded after the tech modules when shutting down gracefully. It is now not possible to unload all the of the calendar modules via "module unload res_calednar.so", but it is still possib…
Branches
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mjordan
Merged revisions 356214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) | 27 lines Fix potential buffer overrun and memory leak when executing "sip show peers" The "sip show peers" command uses a fix sized array to sort the current peers in the peers ao2_container. The size of the array is based on the current number of peers in the container. However, once the size of the array is…
Branches
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seanbright
Make 'iax2 show callnumber usage' output make sense when an IP is passed in. ........ Merged revisions 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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kmoore
Add missing newline to ccss state change notification Move along, nothing to see here...
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seanbright
Remove spurious warning when 'qualifyfreqnotok' is set successfully. (closes issue ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512) ........ Merged revisions 355997 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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seanbright
This was a LOG_NOTICE, so roll it back. ........ Merged revisions 355952 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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seanbright
Change some debug messages from LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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seanbright
Add some boilerplate documentation for IAXVAR and IAXPEER. ........ Merged revisions 355904 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Branches
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seanbright
Set the length of the ast_sockaddr, so that we can set it's port later. Without this, the call to ast_sockaddr_set_port a few lines later is a noop. ........ Merged revisions 355901 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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pabelanger
Blocked revisions 355839 ........ Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
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pabelanger
Revert commit
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pabelanger
Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 355839 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Branches
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alecdavis
push 'outgoing' flag from sig_XXX up to chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ .......…
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seanbright
Don't allow trunkfreq to be greater than 1000ms. ........ Merged revisions 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Branches
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seanbright
Pass the correct value to ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but this value is in milliseconds while ast_timer_set_rate() expects the rate argument to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead. With a default of 20ms, this change makes IAX2 send trunk packets every 20ms instead of every 50ms. Tracked down by myself and Bob Wienholt. ........ …
Branches
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mmichelson
Fix regressions with regards to route-set creation on early dialogs. This fixes two main issues: 1. Asterisk would send a CANCEL to the route created by the provisional response instead of using the same destination it did in the initial INVITE. 2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly possible if our outbound INVITE gets forked), then the route set in the 200 OK needs to overwrite the route set in the 1XX response…
Branches
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seanbright
Revert a change to audio_audiohook_write_list that had no affect. When I made this change initially, I was under the false impression that the audiohooks structure remained on the channel after all of the hooks had been detached. This is not the case, ast ast_read takes care of removing the audiohooks structure if the lists are empty. ........ Merged revisions 355622 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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rmudgett
Fix compile problem when old version of libvorbisfile v1.1.2 is used. The principle difference between libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the configure script to detect if libvorbisfile.h declares OV_CALLBACKS_NOCLOSE. * Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile. (closes issue ASTERIS…
Branches
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rmudgett
Fix AMI Monitor action without File header converting channel name into filename. * Fix potential Solaris crash if Monitor application has a urlbase and no fname_base option. ........ Merged revisions 355574 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Branches
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