asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	76b7966606	Merged revisions 183238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183238 \| russell \| 2009-03-19 12:41:39 -0500 (Thu, 19 Mar 2009) \| 1 line Allow the AES API to work. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	af5ec9ba08	2 symbols defined when DEBUG_THREADS git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	2764c2821f	Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 \| dvossel \| 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) \| 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	10529ea029	Merged revisions 183145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183145 \| russell \| 2009-03-19 11:21:56 -0500 (Thu, 19 Mar 2009) \| 1 line Add missing semicolon in exports script. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5cd40194a3	Merged revisions 183123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183123 \| russell \| 2009-03-19 11:13:18 -0500 (Thu, 19 Mar 2009) \| 2 lines Allow the CallerID API to work again. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	abb71e3d55	Merged revisions 183115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183115 \| mmichelson \| 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) \| 14 lines Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use." A user was having an issue where if an outgoing SIP call was canceled, the SIP device would remain in use if we had not received any response to the initial INVITE we sent out. The SIP device would remain in use until the autocongestion timer was exhausted. I tracked down the cause of this to be the section of code I am removing here. I asked several people what the purpose of this code was meant to be, but no one could give me any sort of answer as to why this was here. The person who was having this issue has been using this patch for several months and it has stopped the problems they have had. AST-196 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	12bd92898d	Improve our triggering of a T38 switchover internally when triggered by a received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	10b7b842dc	Fix an issue where a T38 control frame would get dropped. If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	5a30ea385f	allow this module to export everything for now git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	d22073161f	Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	a12ddc0c15	Blocked revisions 182965 via svnmerge ........ r182965 \| jpeeler \| 2009-03-18 15:02:40 -0500 (Wed, 18 Mar 2009) \| 1 line fix typo which broke configure ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182966 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	4437158c71	Blocked revisions 182963 via svnmerge ........ r182963 \| jpeeler \| 2009-03-18 14:57:05 -0500 (Wed, 18 Mar 2009) \| 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue 0011261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d36ca75e47	Fixing a lost symbol in manager.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4f390ec024	Merged revisions 182882 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182882 \| kpfleming \| 2009-03-18 06:31:41 -0500 (Wed, 18 Mar 2009) \| 3 lines fix another symbol namespace issue (reported by Andrew on asterisk-dev) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	a5c2ac4fc2	a few more namespace updates... res_ael_share still needs some work before this can be merged to other release branches git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0bdd99ad64	Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 \| russell \| 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) \| 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	ab3e9ddad1	Merged revisions 182808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182808 \| kpfleming \| 2009-03-17 20:55:22 -0500 (Tue, 17 Mar 2009) \| 5 lines Improve the build system to properly remove unnecessary symbols from the runtime global namespace. Along the way, change the prefixes on some internal-only API calls to use a common prefix. With these changes, for a module to export symbols into the global namespace, it must have both the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182826 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	16fc1993ef	Add support for the "name" option in the CHANNEL() function. Review: http://reviewboard.digium.com/r/199/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	9dc40485f2	Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	77b2123a99	Blocked revisions 182652 via svnmerge ........ r182652 \| qwell \| 2009-03-17 15:13:40 -0500 (Tue, 17 Mar 2009) \| 7 lines Allow dahdichanname to work as advertised. (closes issue #14056) Reported by: dsedivec Patches: load_from_zapata_conf.patch uploaded by dsedivec (license 638) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	bf2895bae8	Fixing CHANGES in rev 182596. Progress DTMF was added into app_dial's D() option. In CHANGES it should have been updated under 1.6.3 rather than 1.6.2. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	e559cae4ec	Option to send DTMF when receiving PROGRESS status The D() option in app_dial is only able to send DTMF after the call has been answered. A progress option has been added to D() to allow DTMF to be sent upon receiving PROGRESS. This allows DTMF to be sent before the call is answered. (closes issue #12123) Reported by: VoipForces Patches: app_dial.c_patch_trunk_valid uploaded by VoipForces (license 419) dtmf_progress.patch uploaded by dvossel (license 671) Tested by: VoipForces, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	9d6ba51d05	Tweak the handling of the frame list inside of ast_answer(). This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	16b9280ba9	correct logic flaw in ast_answer() changes in r182525 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	d11b6386a5	Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	9456bd1db5	Don't include a space before the optional extra text that may follow a help string. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3e22e8bc94	Merged revisions 182449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182449 \| tilghman \| 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) \| 7 lines Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	7cfc92cc32	OPENR2 uses an incorrect string value if the extension delimiter is not present. * Fixed OPENR2 using an incorrect string value if the extension delimiter is not present in the Dial() function. This was fixed for SS7 and PRI in trunk -r172400. * Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others. * Removed trailing whitespace that appeared with OPENR2. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5e256effa7	Update UPGRADE.txt and CHANGES for 1.6.3 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	77a6840fd3	Add MFC/R2 support for chan_dahdi. This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	18ac659dc6	Merged revisions 182281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 \| dvossel \| 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) \| 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	96a699c065	Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution. Previously, FILE() returned one less character than specified, due to the terminating NULL. Both the offset and length parameters now behave identically to the way variable substitution offsets and lengths also work. (closes issue #14670) Reported by: BMC git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0bf4624643	Merged revisions 182208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182208 \| tilghman \| 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) \| 7 lines Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	5308112806	Fix a memory leak in the ast_answer / __ast_answer API call. For a channel that is not yet answered this API call will wait until a voice frame is received on the channel before returning. It does this by waiting for frames on the channel and reading them in. The frames read in were not freed when they should have been. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b4fcc4a098	Change faulty comparison used when announcing average hold minutes and seconds (closes issue #14227) Reported by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	0892cdb958	Remove ast_ prefix from functions which are not public. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	88e3279f83	Merged revisions 181990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181990 \| mmichelson \| 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) \| 35 lines Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	a6734b5a95	Fix an issue with requesting a T38 reinvite before the call is answered. The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	87a8295303	improve a bit of suboptimal code git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	1fba42de1a	Merged revisions 181898 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Just recording the v1.4 change in trunk since it originally came from here. ........ r181898 \| rmudgett \| 2009-03-12 20:19:29 -0500 (Thu, 12 Mar 2009) \| 4 lines Use the correct branch integrated property when generating the version string. Copied the make_version file from Asterisk trunk. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181899 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a1a9006163	Run the macro on the queue member's channel when he answers, not the caller's channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	593d643d24	Merged revisions 181768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 \| mmichelson \| 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) \| 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	86d6cd8a94	Adjust translation table column widths based upon the translation times. Previously, only 5 columns were displayed, and if a translation time exceeded 99,999 useconds, it would be displayed as 0, instead of its actual time. (closes issue #14532) Reported by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman (license 14) Tested by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	815c56369f	Merged revisions 181664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181664 \| file \| 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) \| 2 lines Fix incorrect usage of strncasecmp... I really meant to use strcasecmp. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	e12265e530	Merged revisions 181659-181660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181659 \| file \| 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) \| 8 lines Fix another scenario where depending on configuration the stream would not get read. For custom commands we don't know whether the audio is coming from a stream or not so we are going to have to read the data despite no channels. (closes issue #14416) Reported by: caspy ........ r181660 \| file \| 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) \| 2 lines Fix logic flaw in previous commit. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	a80c5e37af	Merged revisions 181655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181655 \| file \| 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) \| 10 lines Fix issue with streaming MOH failing if nobody is listening. When a music class is setup to actually provide music on hold from a stream we need to constantly read audio from it since it will constantly be providing audio. This is now done despite there being no channels listening to it. (closes issue #14416) Reported by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	499ca8de65	Fix crash when sleep and retries argument was not given to RetryDial application. (closes issue #14647) Reported by: sherpya git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	7b8564ada6	Whitespace chages. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	b782f28fa9	Use the correct branch integrated property when generating the version string git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1ae8e9f3b	Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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