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2009-11-24Add REPLACE & PASSTHRU functions, overhaul of func_strings, fix API docs for ↵Tilghman Lesher
the ast_get_encoded_* functions. * Add REPLACE function, which searches a given variable for a set of characters and replaces each with a given character. * Add PASSTHRU function, which passes a literal string back, like a NoOp for functions. Intent is to be able to specify a literal string to another function that takes a variable name as an argument. * Let the array manipulation functions work with dialplan functions, in addition to variables. This allows the array manipulation functions to modify ASTDB and ODBC backends, assuming the func_odbc configuration has both read and write functions. (closes issue #15223) Reported by: ajohnson Patches: 20091112__issue15223.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-20Revert code in error and include the gcc suggested workaround for the ↵Tilghman Lesher
original problem, while gcc investigates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-20audiohook signal trigger on every status changeDavid Vossel
(issue #14618) Review: https://reviewboard.asterisk.org/r/434/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-15Increase maximum length of language buffersTilghman Lesher
(closes issue #16217) Reported by: dsessions git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-13Display a list of channel variables in each channel-oriented event.Tilghman Lesher
(Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-09Merged revisions 228827 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov 2009) | 8 lines Perform limited bounds checking when destroying ast_mutex_t structures to make sure we don't try to use negative indices. (closes issue #15588) Reported by: zerohalo Patches: 20090820__issue15588.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Fixes for gcc 4.4Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04mmichelson reported a compilation error related to codec bit expansion that ↵Tilghman Lesher
should be resolved with a simple include of frame_defs.h git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Expand codec bitfield from 32 bits to 64 bits.Tilghman Lesher
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04chan_misdn will fail to compile if the redirect_dn member is missingTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03AMI hook interfaceDavid Brooks
This patch, originally submitted by jozza, enables custom modules to send actions to AMI and receive messages from AMI via a hook interface. Included is a simple test module to illustrate the interface. (closes issue #14635) Reported by: jozza Review: https://reviewboard.asterisk.org/r/412/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03This patch adds a sequence field to CDRs that can be combined with the ↵Matthew Nicholson
linkedid or uniqueid field to uniquely identify a CDR. (closes issue #15180) Reported by: Nick_Lewis Patches: cdr-sequence10.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-02Add PacketCable NCS 1.0 support for Docsis/Eurodocsis networksTilghman Lesher
(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
2009-10-30Add an "Asterisk Architecture Overview" section to the doxygen documentation.Russell Bryant
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
2009-10-28Merged revisions 226304 via svnmerge from Tilghman Lesher
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
2009-10-23Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.Richard Mudgett
* 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
2009-10-22Add Asterisk Git HowTo documentation.Leif Madsen
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
2009-10-22SIP TCP/TLS: move client connection setup/write into tcp helper thread, ↵David Vossel
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
2009-10-22Merged revisions 225105 via svnmerge from Tilghman Lesher
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
2009-10-22Add support for calling and called subaddress. Partial support for COLP ↵Richard Mudgett
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
2009-10-21Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ↵Kevin P. Fleming
ast_channel_iterator to use it. This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the case where multiple results need to be returned; OBJ_NODATA mode already was supported). In addition, it converts ast_channel_iterators (only the targeted versions, not the ones that iterate over all channels) to use this method. During this work, I removed the 'ao2_flags' arguments to the ast_channel_iterator constructor functions; there were no uses of that argument yet, there is only one possible flag to pass, and it made the iterators less 'opaque'. If at some point in the future someone really needs an ast_channel_iterator that does not lock the container, we can provide constructor(s) for that purpose. Review: https://reviewboard.asterisk.org/r/379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Merged revisions 224931 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-17Remove unnecessary typedefTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-15Create an API for adding an optional time unit onto the ends of time periods.Tilghman Lesher
Two examples of its use are included, and the usage could be expanded in some cases into certain configuration options where time periods are specified. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13Fix some doxygen format problems and trim trailing whitespace.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13Fix handling of notification calls w/ the dialing apiTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-08Remove global variable that makes dlopen unhappyTerry Wilson
This isn't the best way to do this, but it is the easiest. There are some limitations that are going to need to be addressed at some point with reloads and when I (or someone else) work on that, then the API can be updated to handle passing the private config data that the calendar tech modules need in a better way as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-08Merged revisions 222878 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-08fixes an ast_netsock_list memory leak.David Vossel
ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-07Deadlock in channel masquerade handlingDavid Vossel
Channels are stored in an ao2_container. When accessing an item within an ao2_container the proper locking order is to first lock the container, and then the items within it. In ast_do_masquerade both the clone and original channel must be locked for the entire duration of the function. The problem with this is that it attemptes to unlink and link these channels back into the ao2_container when one of the channel's name changes. This is invalid locking order as the process of unlinking and linking will lock the ao2_container while the channels are locked!!! Now, both the channels in do_masquerade are unlinked from the ao2_container and then locked for the entire function. At the end of the function both channels are unlocked and linked back into the container with their new names as hash values. This new method of requiring all channels and tech pvts to be unlocked before ast_do_masquerade() or ast_change_name() required several changes throughout the code base. (closes issue #15911) Reported by: russell Patches: masq_deadlock_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, atis (closes issue #15618) Reported by: lmsteffan Patches: deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671) Tested by: lmsteffan, dvossel Review: https://reviewboard.asterisk.org/r/387/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-06Recorded merge of revisions 222152 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Use rtp properties instead of adding a callbackTerry Wilson
Thanks, Josh. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Merged revisions 221086 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-26Allow AES to compile, when OpenSSL is not present.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-25Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channelsPhilippe Sultan
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over XMPP to process calls. SendText can be used instead of JabberSend in the context of XMPP based voice channels (chan_gtalk and chan_jingle). (closes issue #12569) Reported by: eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo Review: https://reviewboard.asterisk.org/r/88/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-24Add bridge related dial flags to the bridge appJeff Peeler
Most of the functionality here is gained simply by setting the feature flag on the bridge config. However, the dial limit functionality has been moved from app_dial to the features code and has been made public so both app_dial and the bridge app can use it. (closes issue #13165) Reported by: tim_ringenbach Patches: app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540), modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-23Add Mantis work flow documention.Leif Madsen
This commit adds the doxygen changes that I've made to describe the Mantis work flow documentation for the open source issue tracker. This should make it easier to determine the flow of issues through the issue tracker, and what those statuses mean. (closes issue #15902) Reported by: lmadsen Patches: mantisworkflow.h uploaded by lmadsen (license 10) Review: https://reviewboard.asterisk.org/r/367/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-17Merged revisions 219136 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-16Detect whether we actually have the long double type, before looking for ↵Tilghman Lesher
those functions. (closes issue #15017) Reported by: tzafrir Patches: 20090916__issue15017.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Verify support for wide ODBC character types before using them.Tilghman Lesher
(closes issue #15870) Reported by: nic_bellamy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-08Add doxygen to ast_event_subscribe for the description.Mark Michelson
Most importantly, note that a NULL description will cause a crash, as I just experienced that firsthand. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-08Ensure that the default autoconf CFLAGS are not used.Kevin P. Fleming
A recent change to the configure script that allows the user to specify CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That problem is now corrected. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04Fix trunk breakage.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04make sure canlog is set so we can compile with DEBUG_THREADS enabled on OpenBSDMichiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Added detection DTMF CID without polarity change alert.Doug Bailey
Added detection of DTMF tone energy levels on FXO channels in chan_dahdi monitoring loop so DTMF CID can be detected without the need of a polarity change precursor. (closes issue #9096) Reported by: fleed Patches: 9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819) Tested by: cyberplant, sum, maturs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Merge code associated with AST-2009-006David Vossel
(closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Revert attempt to standardize with _POSIX_C_SOURCE.Tilghman Lesher
This did not function in the way that was intended, causing more compatibility issues than it solved. It is best, therefore, that it be simply removed. (Discussed with kpfleming; agreement to remove was reached.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-02Let's compile again on OpenBSDMichiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-30Various patches, to enable Asterisk to once again compile on Mac OS X.Tilghman Lesher
One note on defining _POSIX_C_SOURCE: while this feature test macro works to require certain behaviors on Linux, it works differently on *BSD platforms to REMOVE certain API calls that are not in the POSIX specification, such as vasprintf(3). Thus, defining it while depending upon vasprintf (and other extensions to the POSIX standard) to be defined is a recipe to ensure that Asterisk is only buildable on Linux. Hence, this define which was meant to INCREASE portability, effectively ensures the opposite. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-30If lua is detected with the lua5.1 prefix (or not), adjust the include path ↵Tilghman Lesher
accordingly. Based upon feedback to a release announcement on the -users list. See http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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