asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	40dfab583e	Revert 225169, as this doesn't account for the possibility of a list of frames. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	758ed8d437	Isolate the frame returned from ast_translate(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225169 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6e8a455534	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/branches/1.4@225105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8699a5f158	Suffix is not needed for a match git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	bedd6eb8a4	IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	9d65850202	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/branches/1.4@224931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0db2d51ac1	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/branches/1.4@224855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	7de8f53607	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/branches/1.4@224773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	dd9837bba0	Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@224670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	926a033bf9	Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@224565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	7f84021814	Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@224330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	c3501b93e1	Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (in issue 0014292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@224260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jean Galarneau	7499289537	Fix PRI timer T309 operation git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	e3464ac40a	Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	0a226d933f	Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover. (issue #16025) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	6429db49ba	Remove a duplicate ao2_iterator_destroy(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	2f0e76dc39	Remove some unnecessary code. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	f949dd5b17	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	99a8ffb52c	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/branches/1.4@223225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a9317f6cbe	Fix potential memory leak in app_dial.c git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a6e33cd544	fixes sip registration using authuser in user.conf	17 years ago
David Vossel	7d5c81565a	'auth=' did not parse md5 secret correctly (closes issue https://issues.asterisk.org/view.php?id=15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ff6a5575ad	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/branches/1.4@222878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	3e5979a040	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/branches/1.4@222877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	fd238638a0	Fix memory leak if chan_misdn config parameter is repeated. Memory leak when the same config option is set more than once in an misdn.conf section. Why must this be considered? Templates! Defining a template with default port options and later adding to or overriding some of them. Patches: memleak-misdn.patch JIRA ABE-1998 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	7d2cc86d06	chan_misdn.c:process_ast_dsp() memory leak misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	9cc4a5b792	crash on transfer handle_invite_replaces() attempts to uplock a pvt's owner channel without first verifing that it exists. (issue #16027) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	54faffa07f	Add missing unlock(s) in dahdi_read (two cases in trunk) (closes issue #15683) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	7c3d6f732c	Fix potential crash when entire span request is received. The variable index used in this scenario for accessing the dahdi_pvts was wrong and was most likely copied from the several other places it is used correctly. (closes issue #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch uploaded by tsearle (license 373) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	2ad7cb7e87	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/branches/1.4@222152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	dfb8d75f23	Removes unnecessary unlock, clarifies a memcpy. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222026 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	012b1bc180	Ensure the result of the hash function is positive. Negative array offsets suck. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	366ef836c1	Fix a bunch of off-by-one errors git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	ea14c40ae1	Occasionally losing use of B channels in chan_misdn. I have not been able to reproduce the problem of losing channels. However, I have seen in the code a reentrancy problem that might give these symptoms. The reentrancy patch does several things: 1) Guards B channel and B channel structure allocation. 2) Makes the B channel structure find routines more precise in locating records. 3) Never leave a B channel allocated if we received cause 44. The last item may cause temporary outgoing call problems, but they should clear when the line becomes idle. (closes issue #15490) Reported by: slutec18 Patches: issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: rmudgett, slutec18 (closes issue #15458) Reported by: FabienToune Patches: issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: FabienToune, rmudgett, slutec18 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	ae49400957	Use unsigned ints for portinuri flags. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	fe4b70c4f5	Make portinuri a bitfield. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	050d830ec2	Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthias Nick	c179867fb4	changed the prototype definition of csv_quote git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	727d91d733	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/branches/1.4@221200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthias Nick	ebba623307	added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthias Nick	e66c485660	check bounds - prevents for buffer overflow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	96564de25e	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/branches/1.4@221086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	5d982dda69	Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked. (closes issue #15965) Reported by: atis Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96) Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@220907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a0bc561b9e	Reduce CPU usage related to building a peer merely for devicestates. This fixes a 100% CPU problem in the SIP driver, found by profiling the driver while the problem was occurring. (closes issue #14309) Reported by: pkempgen Patches: 20090924__issue14309.diff.txt uploaded by tilghman (license 14) Tested by: pkempgen, vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@220873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	913ba17b12	When selecting DONT_OPTIMIZE in menuselect, explicitly pass -O0 to the compiler so we override any default optimization levels for a particular install. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@220717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5cf26dcdac	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/branches/1.4@220288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	0b21ae08e1	Resolve parallel build warnings. Reported by Klaus Darilion on the asterisk-dev mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@220213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	e8544f5919	Remove the remaining bashisms in the Makefile/mkpkgconfig git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@220099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	0005d493c5	mkpkgconfig does not need bash so make it use /bin/sh This fixes building on all systems that don't have bash at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on #asterisk-dev git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@220027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f4837ecbef	When IMAP variables were changed during a reload, Voicemail did not use the new values. This change introduces a configuration version variable, which ensures that connections with the old values are not reused but are allowed to expire normally. (closes issue #15934) Reported by: viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by tilghman (license 14) Tested by: viniciusfontes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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