asterisk

Commit Graph

Author	SHA1	Message	Date
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	17 years ago
Tilghman Lesher	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	17 years ago
Kevin P. Fleming	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	17 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	17 years ago
Kevin P. Fleming	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	17 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	17 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	17 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	17 years ago
Russell Bryant	1ebf7767d0	Merged revisions 225171 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225171 \| russell \| 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009) \| 2 lines Revert 225169, as this doesn't account for the possibility of a list of frames. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225172 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	9fbb9d0b6c	Merged revisions 225169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225169 \| russell \| 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009) \| 2 lines Isolate the frame returned from ast_translate(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225170 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 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	17 years ago
Tilghman Lesher	77031501a5	Merged revisions 224855 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 \| tilghman \| 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) \| 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	b518c7b9a7	Merged revisions 224773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 \| file \| 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) \| 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c80715706e	Remove unnecessary typedef git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c74a2d0b45	Create an API for adding an optional time unit onto the ends of time periods. 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	17 years ago
Russell Bryant	72934ead4b	Merged revisions 223485-223486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223485 \| russell \| 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) \| 6 lines Don't use data outside of its scope. The purpose of this code was to have a hangup frame put on the list of deferred frames. However, the code that read the hangup frame was outside of the scope of where the hangup frame was declared. ........ r223486 \| russell \| 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) \| 2 lines Remove some unnecessary code. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	0d4726b0a2	Merged revisions 223225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223225 \| mnicholson \| 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) \| 8 lines Signal timeouts by returning AST_CONTROL_RINGING when originating calls. (closes issue #15104) Reported by: nblasgen Patches: manager-timeout1.diff uploaded by mnicholson (license 96) Tested by: nblasgen, mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	dd50b9e8b5	Merged revisions 222878 via svnmerge from 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	17 years ago
David Vossel	db7b4ec65e	fixes an ast_netsock_list memory leak. 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	17 years ago
David Vossel	9456ab2724	Deadlock in channel masquerade handling 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	17 years ago
Kevin P. Fleming	1c9fe00920	Recorded merge of revisions 222152 via svnmerge from 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	17 years ago
Kevin P. Fleming	20743ec07d	Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c1c25181af	Merged revisions 221970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221970 \| tilghman \| 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) \| 2 lines Ensure the result of the hash function is positive. Negative array offsets suck. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221971 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ba10edfcac	Initialize a variable that we check immediately upon startup. (closes issue #15973) Reported by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	bd7ca4b764	One more off-by-one in trunk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8c7b3cf738	Merged revisions 221776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221776 \| tilghman \| 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) \| 2 lines Fix a bunch of off-by-one errors ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	19ba91cd22	Remove ability to control T.38 FAX error correction from udptl.conf. chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	10ce6cd757	Use rtp properties instead of adding a callback Thanks, Josh. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	865daf4858	Merged revisions 221086 via svnmerge from 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	17 years ago
Tilghman Lesher	a4ece92018	Merged revisions 221200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221200 \| tilghman \| 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) \| 7 lines Avoid a potential NULL dereference. (closes issue #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt uploaded by tilghman (license 14) Tested by: kobaz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	01181a27a0	Fix channel reference leak. ast_cel_report_event would geet a reference to the bridged channel. However, certain return paths, such as if CEL was not enabled, would result in a reference leak. All return paths now properly unref the channel. (closes issue #15991) Reported by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	cee8c6cd47	Get rid of annoying and cryptic debug messages. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	d04158f5b1	Eliminate unnecessary include of version.h in manager.c. Including version.h here causes this file to get recompiled after every commit or update, which is not needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	8c30540269	Correct sense of logic test committed in revision 220494. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	aabdc575a5	Don't use hash-based lookups for ast_channel_get_by_name_prefix(). ast_channel_get_full() tries to use OBJ_POINTER to optimize name-based channel lookups, but this will not work properly when the channel's full name was not supplied; for name-prefix searches, there is no value in doing a hash-based lookup, and in fact doing so could result in many channels being skipped. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	17180120bf	Change the default behavior of Set, AGI, and pbx_realtime to 1.6 behavior by default (starting in 1.6.3). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	90746d26f3	fixes tcptls_session memory leak caused by ref count error (closes issue #15939) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/375/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	f150b48bc0	Add bridge related dial flags to the bridge app 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	17 years ago
Tilghman Lesher	1cf5422dc8	Merged revisions 220288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220288 \| tilghman \| 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) \| 6 lines Implicitly sending a progress signal breaks some applications. Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	07f9778f5b	Merged revisions 219653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219653 \| tilghman \| 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) \| 8 lines Really stop the stream, when ast_closestream() is called. (closes issue #15129) Reported by: bmh Patches: 20090918__issue15129.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/372/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	b27a54b8de	Merged revisions 219136 via svnmerge from 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	17 years ago
Tilghman Lesher	3093ccb619	Merged revisions 219023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 \| tilghman \| 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) \| 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	077b44c43f	Merged revisions 218867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218867 \| dbrooks \| 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) \| 13 lines Fixes CID pattern matching behavior to mirror that of extension pattern matching. Pattern matching for extensions uses a type of scoring system, giving values for specificity to each character in the pattern. Unfortunately, this is done character by character, in order. This does lead to some less specific patterns being first in line for matching, but it will usually get the job done. This patch merely brings CID matching to the same level as extension matching. This patch does not attempt to tackle the problem shared by extension matching. (closes issue #14708) Reported by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	3031ca468d	Do not attempt to add a parking extension if an error occurred while reading the configuration. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	1ca9bc4e1e	Check the origination priority for more matches, not the current priority. Found by Pavel Troller on the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	2ee2947a55	Properly terminate the response to the manager Ping action. In passing, correct the formatting of the Timestamp attribute so that there is a space after the colon and before the value. (closes issue #15861) Reported by: Ivan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ad69df830d	Enable turning off the application delimiter warning with the 'dontwarn' option. Suggested on the -dev list, and implemented in an alternate way by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	7348bacf05	Merged revisions 216435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216435 \| mvanbaak \| 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) \| 2 lines make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216506 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	98f18d56b8	Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 \| oej \| 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) \| 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links before any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	6f32731568	make sure 'start' is always initialized. Makes asterisk compile with --enable-dev-mode git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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