asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	49bf540c71	Create iterative method for querying SRV results, and use that for finding AGI servers. (closes issue #14775) Reported by: _brent_ Patches: 20091215__issue14775.diff.txt uploaded by tilghman (license 14) hagi-5.patch uploaded by brent (license 388) Tested by: _brent_ Reviewboard: https://reviewboard.asterisk.org/r/378/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	568c057c4c	Extend max call limit duration from 24.8 days to 292+ million years. If the limit was set past MAX_INT upon answering, the call was immediately hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup). The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been changed to return an int64_t to prevent overflow. Also the reporter suggested adding a message indicating the reason for the call hanging up. Given that the new limit is so much higher, the message (which would only really be useful in the overflow scenario) has been made a debug message only. (closes issue #16006) Reported by: viraptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	ddad718f8e	Note where empty lines should reside in commit messages. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e8a6d2995e	Add pickup event to AMI. Also, fix AMI documentation. (closes issue #16431) Reported by: syspert Patches: 20100112__issue16431.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ecbe7eff7a	Add the TESTTIME() dialplan function, which permits testing GotoIfTime. Specifically, by setting TESTTIME() to a particular date and time, you can test whether a dialplan correctly branches as was intended. This was developed after recent questions on the -users list on how to test their holiday dialplan logic. (closes issue #16464) Reported by: tilghman Patches: 20100112__issue16464.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/458/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	bd2c63a59d	Adding Tilghman's documentation from asterisk-dev to the actual file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	bf06747778	fixes AUDIOHOOK_INHERIT regression During the process of removing an audiohook from one channel and attaching it to another the audiohook's status is updated to DONE and then back to whatever it was previously. Typically updating the status after setting it to DONE is not a good idea because DONE can trigger unrecoverable audiohook destruction events... because of this a conditional check was added to audiohook_update_status to explicitly prevent the audiohook from ever changing after being set to DONE. It was this check that prevented audiohook inherit from work properly though. Now ast_audiohook_move_by_source is treated as a special exception, as the audiohook must be returned to its previous status after attaching it to the new channel. This is only a safe operation because the audiohook's lock is held the entire time, otherwise this could cause trouble. (closes issue #16522) Reported by: corruptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	b70bc21627	fixes test.c compile issue when TEST_FRAMEWORK is not enabled The ast_test_status_update() function is defined in test.h. When TEST_FRAMEWORK is not enabled a macro is defined as a no-op place holder for this function. The macro did not contain the correct number of arguments. This caused a compile error. Much thanks to wdoekes for reporting the issue and supplying the patch! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	386b847075	Merged revisions 237405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237405 \| tilghman \| 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) \| 16 lines Add a flag to disable the Background behavior, for AGI users. This is in a section of code that relates to two other issues, namely issue #14011 and issue #14940), one of which was the behavior of Background when called with a context argument that matched the current context. This fix broke FreePBX, however, in a post-Dial situation. Needless to say, this is an extremely difficult collision of several different issues. While the use of an exception flag is ugly, fixing all of the issues linked is rather difficult (although if someone would like to propose a better solution, we're happy to entertain that suggestion). (closes issue #16434) Reported by: rickead2000 Patches: 20091217__issue16434.diff.txt uploaded by tilghman (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: rickead2000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	2706de850a	Merged revisions 236585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236585 \| seanbright \| 2009-12-28 10:12:08 -0500 (Mon, 28 Dec 2009) \| 7 lines Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces. There was conditional code (based on build platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add a configure-time check for it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ffd9d82472	Allow test_heap.c to compile when AST_DEVMODE is true, but TEST_FRAMEWORK is false git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	73cb2d507b	Unit Test Framework API The Unit Test Framework is a new API that manages registration and execution of unit tests in Asterisk with the purpose of verifying the operation of C functions. The Framework consists of a single test manager accompanied by a list of registered test functions defined within the code. A test is defined, registered, and unregistered from the framework using a set of macros which allow the test code to only be compiled within asterisk when the TEST_FRAMEWORK flag is enabled in menuselect. This allows the test code to exist in the same file as the C functions it intends to verify. Registered tests may be viewed and executed via a set of new CLI commands. CLI commands are also present for generating and exporting test results into xml and txt formats. For more information and use cases please refer to the documentation provided at the beginning of the test.h file. Review: https://reviewboard.asterisk.org/r/447/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ef9be94b35	Change all refererences to 1.6.3 to be 1.8, since that will be the next feature release git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	cf7b67d9d3	Merged revisions 235635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235635 \| jpeeler \| 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) \| 48 lines Correct CDR dispositions for BUSY/FAILED This patch is simple in that it reorders the disposition defines so that the fix for issue 12946 works properly (the default CDR disposition was changed to AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all CDR records are written. The side effects of CDR changes are scary, so I'm documenting the test cases performed to attempt to catch any regressions. The following tests were all performed using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A (Both SIP and features) A calls B A blind transfers to C Hangup C (Both SIP and features) A calls B A attended transfers to C Hangup C A calls B A attended transfers to C (SIP) C blind transfers to A (features) Hangup A All of the test scenario CDRs matched. The following tests were performed just with the patch to ensure proper operation (with unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue #16180) Reported by: aatef Patches: bug16180.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	6b34563778	Add auth_policy option to jabber.conf for auto user registration. The option is global and currently the acceptable values as noted in the sample config are accept or deny. (closes issue #15228) Reported by: lp0 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d4894b3d25	Is it Friday yet? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	26daf50863	Add applications JabberJoin, JabberLeave, JabberSendGroup for XMPP groupchat (closes issue #14352) Reported by: fiddur Patches: trunk-14352-2.diff uploaded by phsultan (license 73) Tested by: fiddur git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	cfd17ef0a6	Move implementation of closefrom(3) from app.c to strcompat.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	aa9ec67f97	OS X does not define MSG_NOSIGNAL, but it does have a socket option SO_NOSIGPIPE. (closes issue #16178) Reported by: oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7e0a2db236	Fix multiple issues with musiconhold, which led to classes not getting destroyed properly. * Classes are now tracked past removal from the core container, and module removal is actively prevented until all references are freed. * A hanging reference stored in the channel has been removed. This could have caused a mismatch and the music state not properly cleared, if two or more reloads occurred between MOH being stopped and MOH being restarted. * In certain circumstances, duplicate classes were possible. * A race existed at reload time between a process being killed and the thread responsible for reading from the related pipe respawning that process. * Several reference counts have also been corrected. At least one could have caused deleted classes to stick around forever, consuming resources. This originally manifested as MOH external processes that were not killed at reload time. (closes issue #16279, closes issue #16207) Reported by: parisioa, dcabot Patches: 20091202__issue16279__2.diff.txt uploaded by tilghman (license 14) Tested by: parisioa, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f46840c107	So apparently, some platforms don't have ffsll(3). The manpage lies; it says that the function is in POSIX, but that's only for ffs(3), not ffsll(3). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f59fe83c56	More 32->64 bit codec conversions. In the process of swapping ULAW to a place in the extended codec space, we found several unhandled cases, where a 32-bit integer was still being used to handle a codec field. Most of these have been fixed with this commit, although there is at least one case (codec_dahdi) which depends upon outside headers to be altered before a conversion can be made. (Fixes AST-278, SWP-459) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b2d115bce9	Formats need to be able to represent all 64 codec bits. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	5ba2b689b2	Another round of UDPTL stack fixes/improvements: 1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL session, so that log/error/debug messages generated by the UDPTL stack can be 'connected' to the endpoint that caused them to be generated. 2) Improve comments (and process) of calculating the far end's maximum IFP size when redundancy mode is in use for error correction. 3) When an IFP larger than the calculated 'far max IFP' size is presented for writing, truncate it rather than putting in the buffer and allowing the buffer to overflow; this will cause the ends to retrain to a lower bit rate that produces IFPs of an appropriate size if possible, and if not possible, the FAX transfer will fail completely. In these cases, it is due to the one endpoint supplying a T38FaxMaxDatagram value that is improperly calculated and is too low to be of use; we have configuration options available to override this behavior. 4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	31848bcdd1	Merged revisions 231614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 \| mnicholson \| 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) \| 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	936a2bd202	Reverted 231616 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	8d1f4fa5ea	Merged revisions 231614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 \| mnicholson \| 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) \| 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0bccc4fbe6	Add REPLACE & PASSTHRU functions, overhaul of func_strings, fix API docs for 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	16 years ago
Tilghman Lesher	b6378e07d7	Revert code in error and include the gcc suggested workaround for the original problem, while gcc investigates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3595fbb70c	audiohook signal trigger on every status change (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	16 years ago
Tilghman Lesher	f4d50dc70d	Increase maximum length of language buffers (closes issue #16217) Reported by: dsessions git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5e2aa190fe	Display a list of channel variables in each channel-oriented event. (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	16 years ago
Matthew Nicholson	88d5fedb34	Merged revisions 228827 via svnmerge from 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	16 years ago
Tilghman Lesher	d5fa4289d0	Fixes for gcc 4.4 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8d1befcbe8	mmichelson reported a compilation error related to codec bit expansion that 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	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6a50e7a031	chan_misdn will fail to compile if the redirect_dn member is missing git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	d87006ca1c	AMI hook interface 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	16 years ago
Matthew Nicholson	7ed425ec80	This patch adds a sequence field to CDRs that can be combined with the 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	16 years ago
Tilghman Lesher	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
Russell Bryant	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
Tilghman Lesher	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
Richard Mudgett	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
Leif Madsen	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
David Vossel	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
Tilghman Lesher	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
Richard Mudgett	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
Kevin P. Fleming	cdd1f9e296	Finish implementaton of astobj2 OBJ_MULTIPLE, and convert 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	16 years ago
Russell Bryant	cd10bd931a	Merged revisions 224931 via svnmerge from 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	16 years ago
Tilghman Lesher	c80715706e	Remove unnecessary typedef git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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