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${ noResults }
18899 Commits (c900eaa7334c7d92d364badefb8d4ca569f81bb5)
Author | SHA1 | Message | Date |
---|---|---|---|
|
c900eaa733 |
Merged revisions 227166 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 lines Fix a bug where an RPID header could be generated with a blank username in the URI. (closes issue #15909) Reported by: kobaz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227167 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
f6827928b0 |
Update extensions.conf.sample file to fix incorrect extensions.
(closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227162 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ede3699c6e |
Merged revisions 227088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 lines Use proper response code when violating Contact ACL's. https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
66579d9d49 |
Add PacketCable NCS 1.0 support for Docsis/Eurodocsis networks
(closes issue #12950) Reported by: alea-soluciones Patches: ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones (license 514) Tested by: alea-soluciones, adomjan, urtho, nahuelgreco git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227049 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
2c4d3b3168 |
SIP channel name uniqueness
SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226974 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
d279260f51 |
SIP channel name uniqueness
SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226973 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
8d64231c62 |
Adding external reference for doxygen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226970 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
7a17d87740 |
Merged revisions 226889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226890 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
6406f39594 |
DAHDI ISDN channel names will not allow device state to work. (Interim solution.)
Since ISDN works like SIP and not analog ports in regard to devices, the device state based on the ISDN channel number could not work. This has not been an issue until the advent of PTMP NT mode. Previously, ISDN lines were used as trunks and did not have to keep track of specific devices. As an interim solution until device states are properly implemented, the channel name is being changed to the following format to use the generic device state support: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> Dialplan hints would thus be: exten => xxx,hint,DAHDI/i2/5551212 This will work with the following restrictions: * The number of devices/phones cannot exceed the number of B channels. (i.e., BRI has 2) * Each device/phone can only have one number. No shared MSN's. * The phones/devices probably should not use subaddressing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226882 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
14a8c042cd |
Merged revisions 226811 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) | 8 lines Don't allow two separate instances of safe_asterisk when restarting from the init script. (closes issue #14562) Reported by: davidw Patches: Initially 20091022__issue14562.diff.txt uploaded by tilghman (license 14) Modified to 20091030__Issue14562_diff.txt uploaded by davidw (license 780) Tested by: davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226812 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
d107a0e3cf |
Blocked revisions 226736 via svnmerge
........ r226736 | dvossel | 2009-11-02 09:31:02 -0600 (Mon, 02 Nov 2009) | 5 lines fixes crash on iterator_destroy on uninitialized iterator (closes issue #16162) Reported by: krn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226748 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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49fc72fc7c |
Blocked revisions 226688 via svnmerge
........ r226688 | dvossel | 2009-11-02 09:16:30 -0600 (Mon, 02 Nov 2009) | 5 lines changes calltoken debug messages from LOG_NOTICE to LOG_DEBUG like they are supposed to be (closes issue #16144) Reported by: aragon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226689 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
93e43578ec |
This patch adds support for a draft proposal for adding Q.850 reason headers to sip messages.
(closes issue #13385) Reported by: adomjan Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by adomjan (license 487) sip-q850-hangupcause1.diff uploaded by mnicholson (license 96) Tested by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
7fbd314a88 |
Cleanup some flags on DAHDI PRI channel hangup.
* Cleanup some flags on DAHDI PRI channel hangup. (sig_pri split) * Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls. * Remove some unused flags since sig_pri was split. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226648 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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844a01b27e |
Add an "Asterisk Architecture Overview" section to the doxygen documentation.
This is a side project I've been poking at this week. The intent is to discuss Asterisk architecture in a top down fashion to help new developers understand how Asterisk is put together. There is a ton of stuff to write about, so this will just continue to evolve over time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226606 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b9c370da86 |
Merged revisions 226531 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226532 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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64e8fb465b |
Doxygen documentation update
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5bd895b24b |
remove empty awk pattern (//)
Solaris 10 nawk doesn't lthe empty pattern ike '//' for 'always'. Just remove that. No pattern at all always matches. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226453 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5524f0ab11 |
Merged revisions 226382 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226384 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6b6fee6c95 |
Merged revisions 226377 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines Update CALLINGSUBADDR channel variable documentation. (closes issue #15734) Reported by: alecdavis Patches: channelvariables.tex.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226378 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
3afd1409d1 |
Merged revisions 226304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines Fix documentation (pointed out by TheDavidFactor on #-dev) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226305 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
3b6d052ac7 |
Remove extra cleanup in case we have more than one Asterisk.
/var/run would be cleaned on startup on most systems anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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56a6df50cc |
another variation of the upstart script
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
891e3df1c6 |
Adding compile time flags for Snow Leopard, Leopard and some other animals
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226184 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c6a34a201e |
Merged revisions 226138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines Manager output is not always NULL-terminated, so force a NULL at the end of the filestream. (closes issue #15495) Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded by tilghman (license 14) Tested by: pdf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ba8176dc52 |
Don't prepend the URI prefix to the post directory
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226099 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5825f68e8b |
Add support for receiving unsolicited MWI NOTIFY messages.
This change adds a configuration option to SIP peers, unsolicited_mailbox, which configures a virtual mailbox to use for received new/old MWI information. This virtual mailbox can then be used by any device supporting MWI. (closes issue #13028) Reported by: AsteriskRocks Patches: bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7e29f0da44 |
detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi
* Set OSARCH to linux-gnu even if host_os is linux-gnueabi * When checking if we are Linux, check OSARCH rather than host_os The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is tested for the value of 'linux-gnu' in one or two places in the tree. This patch also fixes the check libcap to check for $OSARCH rather than $host_os . See also: http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226018 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ea8b54fb9d |
Fix building in REF_DEBUG mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225956 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ad7a9fb8e8 |
Correct broken logic from revision 225405.
The code committed in revision 225405 was broken; instead of removing the unreference code, the logic used to decide when to do it should have been reversed. This patch corrects the situation, and makes reference counting work properly again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225955 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ec0a1882c9 |
ACL check not present for verifying SIP INVITEs
The ACL check in check_peer_ok was missing and has now been restored. The missing check allowed for calls to be made on prohibited networks where an ACL was defined in sip.conf and the allowguest option was set to off. See the AST security advisory below for more information. Merge code associated with AST-2009-007. (closes issue #16091) Reported by: thom4fun git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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71452322a2 |
Make conditionals create previous code when libpri/ss7 are present.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225872 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2736168a6b |
span numbers in pri debug / error messages
Prefix PRI trace messages with the span number. This makes the trace readable even when you have a multi-port device. (closes issue #15054) Reported by: tzafrir Patches: dahdi_pri_debug_spannum.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225836 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e5a57959eb |
Re-arange code a bit to build in dev-mode without ss7
No change of functionality here. Just localized a variable and indented code into blocks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225803 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d36cecd578 |
Make chan_dahdi build even without PRI / SS7
(Note: still some strange build warnings without SS7 in dev-mode) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225767 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
fb0196fce6 |
Improve performance of pedantic mode dialog searching in chan_sip.
This patch changes chan_sip to use the new astobj2 OBJ_MULTIPLE iterator support to make pedantic mode dialog searching in find_call() not require a linear search of all dialogs in the list of dialogs. This patch does *not* change the dialog matching logic (more on that later), just improves the searching performance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225727 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
cff6d02b53 |
Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.
* Added handling of received HOLD/RETRIEVE messages and the optional ability to transfer a held call on disconnect similar to an analog phone. * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP. Will reroute/deflect an outgoing call when receive the message. Can use the DAHDISendCallreroutingFacility to send the message for the supported switches. * Added ability to send/receive keypad digits in the SETUP message. Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension]) Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)} * Added support for BRI PTMP NT mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225692 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
52a2ded0cd |
Optionally build and install the sample AGIs in the agi/ directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225690 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
2208fb171b |
Fixes an iterator memory leak and uninitialized memory
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225650 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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673eadaa3a |
Merged revisions 225581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on every build. For some reason the menuselect.makeopts file was listed as PHONY in the Makefile, resulting in 'make' needing to rebuild it for every build. This then resulted in the embedded module rules being rebuilt on every build, which can be slow and is unnecessary. This patch fixes the problem by properly allowing 'make' to know when the menuselect.makeopts file needs to be rebuilt (defining the proper dependencies). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
6cb07e7030 |
Update README documentation.
Update the README documentation to correctly describe which CLI command you should use when attempting to get help from the CLI. (closes issue #16064) Reported by: thedavidfactor Patches: readme.patch uploaded by thedavidfactor (license 903) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225515 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b544c53778 |
Merged revisions 225484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines Clean valgrind output by suppressing false errors. Update valgrind.txt documentation and add valgrind.supp file in order to allow those who are creating valgrind output to have less false errors in the logfile. (closes issue #16007) Reported by: atis Patches: valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp uploaded by atis (license 242) Tested by: atis, amorsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225485 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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681ec86837 |
Add Asterisk Git HowTo documentation.
Added documentation on how to create a local git repository from SVN. This documentation was added via doxygen. (closes issue #15814) Reported by: tzafrir Patches: git-asterisk-howto uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225483 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
63473616da |
Search for the subaddress only within the extension section of the dial string.
Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension]) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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776a14386a |
SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.
What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225445 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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117d3fd910 |
Add the programs in utils/ to menuselect.
Nothing in utils/ is now built by default except for astcanary. Review: https://reviewboard.asterisk.org/r/353/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225440 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d9f72c1893 |
Permit storage of voicemail secrets in a separate file, located within the spool directory.
(closes issue #14276) Reported by: klaus3000 Patches: app_voicemail.c-svn-trunk-r214898.txt uploaded by klaus3000 (license 65) Tested by: jamesgolovich git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225406 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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be796e703a |
Fix a refcount error introduced by yesterday's OBJ_MULTIPLE commit.
When an object is being unlinked from its container *and* being returned to the caller, we do not want to decrement the reference count after unlinking it from the container, as the reference that the container held is what we are returning to the caller... and if it was the only remaining reference to the object, that could result in the object being destroyed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225405 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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496282194c |
Merged revisions 225105 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225360 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
1174a61612 |
Add support for calling and called subaddress. Partial support for COLP subaddress.
The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing "desk to desk" between offices each with an asterisk box over the ISDN should then be possible, without a whole load of DDI numbers required. (closes issue #15604) Reported by: alecdavis Patches: asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585) Some minor modificatons were made. Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/405/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |