asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin P. Fleming	dcefa607e5	Merged revisions 209256 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 \| kpfleming \| 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) \| 10 lines Make T.38 switchover in ReceiveFAX synchronous. In receive mode, if the channel that ReceiveFAX is running on supports T.38, we should always attempt to switch T.38, rather than listening for an incoming CNG tone and only triggering on that. The channel may be using a low-bitrate codec that distorts the CNG tone, the sending FAX endpoint may not send CNG at all, or there could be a variety of other reasons that we don't detect it, but in all those cases if T.38 is available we certainly want to use it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	af177fd104	Blocked revisions 209235 via svnmerge ........ r209235 \| mmichelson \| 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) \| 5 lines Gracefully handle malformed RTP text packets. AST-2009-004 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	6d177693e0	Merged revisions 209098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 \| dbrooks \| 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) \| 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7f81823913	Merged revisions 209197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209197 \| mmichelson \| 2009-07-27 15:11:42 -0500 (Mon, 27 Jul 2009) \| 9 lines Honor channel's music class when using realtime music on hold. (closes issue #15051) Reported by: alexh Patches: 15051.patch uploaded by mmichelson (license 60) Tested by: alexh ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	881cdb00e5	Merged revisions 209132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209132 \| mmichelson \| 2009-07-27 12:50:04 -0500 (Mon, 27 Jul 2009) \| 24 lines Merged revisions 209131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209131 \| mmichelson \| 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) \| 18 lines Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	7351db2bd1	Just replacing typos "recieved" with "received". From issue #15360, forgot to apply to trunk and other branches. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	a1cb9549be	Merged revisions 209056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 \| kpfleming \| 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) \| 10 lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and underscore-variants to sub-makes. During the recent Makefile improvements I made, it seemed the 'make' was automatically carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so I removed the explict export of them. However, there are some circumstances where make does this, and some where it does not, so I've brought them back to ensure they are always exported. I also removed an extraneous double setting of _ASTLDFLAGS on *BSD platforms. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	ab117ba86d	Merged revisions 208924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208924 \| jpeeler \| 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009) \| 9 lines Merged revisions 208923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208923 \| jpeeler \| 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) \| 2 lines Fix logic errors from 208746 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	420cfcc16a	Blocked revisions 208886 via svnmerge ........ r208886 \| mvanbaak \| 2009-07-26 16:00:52 +0200 (Sun, 26 Jul 2009) \| 2 lines add OpenBSD to the install_prereq script ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	aa3928f100	Blocked revisions 208848 via svnmerge ........ r208848 \| mvanbaak \| 2009-07-25 14:28:38 +0200 (Sat, 25 Jul 2009) \| 2 lines libxml2-dev is needed as well by default. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	ed6555a37c	Blocked revisions 208813 via svnmerge ........ r208813 \| mvanbaak \| 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) \| 10 lines add default alias reload to run module reload. Requiring 'module reload' to reload everything, including core etc makes russell very unhappy. The default configuration already loads the 'friendly' aliases template. Added 'reload=module reload' to that template. Also removed the comment in main/cli.c that reload should come back. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	bf0a2c9fa5	Merged revisions 208749 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208749 \| jpeeler \| 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) \| 13 lines Merged revisions 208746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208746 \| jpeeler \| 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) \| 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a564e9b944	Blocked revisions 208622 via svnmerge ........ r208622 \| mmichelson \| 2009-07-24 14:24:28 -0500 (Fri, 24 Jul 2009) \| 16 lines Don't impose an arbitrary limit on member lines in queues.conf I know what some of you are thinking: "UGH! Mark, why are you using ast_strdup and ast_free for the string when you can just use ast_strdupa and let the memory free itself?! Have the bats been chewing on your brain again?" Based on past experiences, I don't like using ast_strdupa inside a loop. It's a good way to potentially exhaust stack space. Also, since this only happens when reloading queues, I don't think that heap allocations and frees are going to be a huge problem. (closes issue #15559) Reported by: amorsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	765d10231b	Merged revisions 208593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208593 \| russell \| 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) \| 14 lines Merged revisions 208592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208592 \| russell \| 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) \| 7 lines Do not log an ERROR if autoservice_stop() returns -1. This does not indicate an error. A return of -1 just means that the channel has been hung up. (reported in #asterisk-dev) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	6aa63436ab	Merged revisions 208588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208588 \| mmichelson \| 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) \| 16 lines Merged revisions 208587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208587 \| mmichelson \| 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) \| 10 lines Only send a BYE when hanging up a channel that is up. For cases where Asterisk sends an INVITE and receives a non 2XX final response, Asterisk would follow the INVITE transaction by immediately sending a BYE, which was unnecessary. (closes issue #14575) Reported by: chris-mac ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	f43a65fd21	Merged revisions 208548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 \| kpfleming \| 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) \| 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	84671eef56	Blocked revisions 208542 via svnmerge ........ r208542 \| mvanbaak \| 2009-07-24 16:35:49 +0200 (Fri, 24 Jul 2009) \| 13 lines use aptitude for debian based systems The function to check wether we need to install packages was using dpkg-query which was gives wrong output on Debian 5 Also, the apt-get has been replaced with aptitude because aptitude is now the preferred way to handle packages on Debian (closes issue #15570) Reported by: mvanbaak Patches: 2009072400_installprereq-aptitude.diff uploaded by mvanbaak (license 7) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	560384cf09	Use correct formatting for T.38 change note in UPGRADE.txt git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	791d4f0478	Merged revisions 208464 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 \| kpfleming \| 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) \| 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum IFP size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending our MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	db6c757a3d	Merged revisions 208388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208388 \| mmichelson \| 2009-07-23 14:34:49 -0500 (Thu, 23 Jul 2009) \| 24 lines Merged revisions 208386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208386 \| mmichelson \| 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) \| 17 lines Fix a problem where a 491 response could be sent out of dialog. This generalizes the fix for issue 13849. The initial fix corrected the problem that Asterisk would reply with a 491 if a reinvite were received from an endpoint and we had not yet received an ACK from that endpoint for the initial INVITE it had sent us. This expansion also allows Asterisk to appropriately handle an INVITE with authorization credentials if Asterisk had not received an ACK from the previous transaction in which Asterisk had responded to an unauthorized INVITE with a 407. (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	8d49fb4502	Merged revisions 208383 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208383 \| jpeeler \| 2009-07-23 14:21:50 -0500 (Thu, 23 Jul 2009) \| 12 lines Merged revisions 208380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208380 \| jpeeler \| 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) \| 6 lines Only set the priindication setting when not performing a reload (closes issue #14696) Reported by: fdecher ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a9ad08042d	Merged revisions 208314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208314 \| mmichelson \| 2009-07-23 11:29:37 -0500 (Thu, 23 Jul 2009) \| 9 lines Merged revisions 208312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208312 \| mmichelson \| 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) \| 3 lines Remove inaccurate XXX comment. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	0bf989f1c0	Blocked revisions 208267 via svnmerge ........ r208267 \| jpeeler \| 2009-07-23 10:59:44 -0500 (Thu, 23 Jul 2009) \| 13 lines Fix sending of interface identifier unconditionally in sig_pri The wrong logic was being used in chan_dahdi to convert a sig_pri_chan to the proper libpri channel number. The most significant bit must only be set only when trunk groups are being used. (closes issue #15452) Reported by: alecdavis Patches: bug15452.patch uploaded by jpeeler (license 325) Tested by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	0a6ccac217	Merged revisions 208263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208263 \| mmichelson \| 2009-07-23 10:46:34 -0500 (Thu, 23 Jul 2009) \| 15 lines Merged revisions 208262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208262 \| mmichelson \| 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) \| 8 lines Properly handle 183 responses which do not contain an SDP. (closes issue #15442) Reported by: ffloimair Patches: 15442.patch uploaded by mmichelson (license 60) Tested by: tkarl, ffloimair ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	444475f1cc	Blocked revisions 208155 via svnmerge ........ r208155 \| jpeeler \| 2009-07-22 17:42:33 -0500 (Wed, 22 Jul 2009) \| 5 lines Reset the fax buffers back to default settings regardless of signaling in use - Pointed out by Matt F. Also in the case of not using a signaling module, set the law back to the default as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	fbe102e8f1	Blocked revisions 208113 via svnmerge ........ r208113 \| qwell \| 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) \| 9 lines Restore an int declaration on PPC platforms. This x is one crafty little bugger... It was used for 2 different things (one of which was only done on PPC) in 1.4. One of the uses were removed in trunk, and with it went the declaration. (closes issue #14038) Reported by: ffloimair ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	5c18c82487	Blocked revisions 207950 via svnmerge ........ r207950 \| jpeeler \| 2009-07-21 17:51:47 -0500 (Tue, 21 Jul 2009) \| 7 lines Do not dial digits when none were specified for sig_pri based calls (closes issue #15524) Reported by: elguero Patches: pri-sig-no-dest-set.patch uploaded by elguero (license 37) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	7d73264772	Merged revisions 207946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207946 \| tilghman \| 2009-07-21 17:45:32 -0500 (Tue, 21 Jul 2009) \| 15 lines Merged revisions 207945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207945 \| tilghman \| 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) \| 8 lines Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional). This change makes URIENCODE and QUOTE behave similarly, since the documentation states that the argument is not optional, for both. (closes issue #15439) Reported by: pkempgen Patches: 20090706__issue15439.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	c62d1b0b58	Blocked revisions 207902 via svnmerge ........ r207902 \| jpeeler \| 2009-07-21 17:02:25 -0500 (Tue, 21 Jul 2009) \| 2 lines Fix my_is_off_hook to check rxbits only for FXS signaling ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	4ab9bff204	Merged revisions 207854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207854 \| jpeeler \| 2009-07-21 15:26:02 -0500 (Tue, 21 Jul 2009) \| 16 lines Merged revisions 207827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207827 \| jpeeler \| 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) \| 9 lines Wait for wink before dialing when using E&M wink signaling There was already code for other signaling types in dahdi_handle_event to handle dialing if a dial operation dial string was present. Simply add SIG_EMWINK to the list. (closes issue #14434) Reported by: araasch ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	ab6510ebf1	Revert r207636, this approach could potentially block for an unacceptable amount of time. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	51101e961b	Merged revisions 207723 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207723 \| mmichelson \| 2009-07-21 09:29:40 -0500 (Tue, 21 Jul 2009) \| 11 lines Merged revisions 207714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207714 \| mmichelson \| 2009-07-21 09:26:00 -0500 (Tue, 21 Jul 2009) \| 5 lines Document default timeout for AMI originations. AST-224 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207725 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	69255bd210	Merged revisions 207680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207680 \| kpfleming \| 2009-07-21 08:28:04 -0500 (Tue, 21 Jul 2009) \| 18 lines Merged revisions 207647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207647 \| kpfleming \| 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) \| 12 lines Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker after all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided either in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	55a51b9194	Wait for wink before dialing when using E&M wink signaling This patch adds a new dahdi_wait function to specifically wait for the wink event. If the wink is not eventually received the channel is hung up. (closes issue #14434) Reported by: araasch Patches: emwinkmod uploaded by araasch (license 693) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	935f33e481	Merged revisions 207424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207424 \| mmichelson \| 2009-07-20 14:48:12 -0500 (Mon, 20 Jul 2009) \| 39 lines Merged revisions 207423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207423 \| mmichelson \| 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) \| 33 lines Answer video SDP offers properly when videosupport is not enabled. Copied from Review board: In issue 12434, the reporter describes a situation in which audio and video is offered on the call, but because videosupport is disabled in sip.conf, Asterisk gives no response at all to the video offer. According to RFC 3264, all media offers should have a corresponding answer. For offers we do not intend to actually reply to with meaningful values, we should still reply with the port for the media stream set to 0. In this patch, we take note of what types of media have been offered and save the information on the sip_pvt. The SDP in the response will take into account whether media was offered. If we are not otherwise going to answer a media offer, we will insert an appropriate m= line with the port set to 0. It is important to note that this patch is pretty much a bandage being applied to a broken bone. The patch only helps for situations where video is offered but videosupport is disabled and when udptl_pt is disabled but T.38 is offered. Asterisk is not guaranteed to respond to every media offer. Notable cases are when multiple streams of the same type are offered. The 2 media stream limit is still present with this patch, too. In trunk and the 1.6.X branches, things will be a bit different since Asterisk also supports text in SDPs as well. (closes issue #12434) Reported by: mnnojd Review: https://reviewboard.asterisk.org/r/311 Review: https://reviewboard.asterisk.org/r/313 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	784e78e526	Merged revisions 207361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207361 \| russell \| 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009) \| 16 lines Merged revisions 207360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207360 \| russell \| 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) \| 9 lines Only do the chan->fdno check in ast_read() in a developer build. I changed this check to only happen in a dev-mode build. I also added a comment explaining what is going on. I also made it so that detection of this situation does not affect ast_read() operation. (closes issue #14723) Reported by: seadweller ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	c305a6d0a3	Merged revisions 145293,158010 from https://origsvn.digium.com/svn/asterisk/branches/1.4 to make merging easier. These changes are already on trunk. ................ r145293 \| rmudgett \| 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) \| 54 lines channels/chan_misdn.c channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 \| rmudgett \| 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) \| 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 \| crichter \| 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) \| 9 lines improved helptext of misdn_set_opt. ........ r142181 \| rmudgett \| 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) \| 1 line Cleaned up comment ........ r138738 \| rmudgett \| 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) \| 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. ................ r158010 \| rmudgett \| 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) \| 9 lines Merged revision 157977 from https://origsvn.digium.com/svn/asterisk/team/group/issue8824 ........ Fixes JIRA ABE-1726 The dial extension could be empty if you are using MISDN_KEYPAD to control ISDN provider features. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207286 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	08fb833859	Merged revisions 207156 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207156 \| jpeeler \| 2009-07-17 14:37:38 -0500 (Fri, 17 Jul 2009) \| 14 lines Merged revisions 207155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207155 \| jpeeler \| 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) \| 7 lines Fix format specifier to print out an unsigned long long. Yep, it's even ifdefed out code. But it made it to the RR list... (closes issue #14726) Reported by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	21526941cb	Merged revisions 207095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207095 \| jpeeler \| 2009-07-17 14:16:35 -0500 (Fri, 17 Jul 2009) \| 2 lines Update some missing allowed options for overlapdial ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	5f6fa4990f	Merged revisions 207029 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 \| dvossel \| 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) \| 6 lines sip option flags handled incorrectly (closes issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, Takehiko_Ooshima ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	e577092f8e	Blocked revisions 206998 via svnmerge ........ r206998 \| jpeeler \| 2009-07-17 12:02:44 -0500 (Fri, 17 Jul 2009) \| 14 lines Fix segfault in sig_analog when using callwaiting, respect callwaiting options Sig_analog handles allocating the sub channel for callwaiting, so no longer try to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as allocated upon success of the alloc_sub callback, which was responsible for the segfault. Also, the callwaiting and callwaitingcallerid options were being unconditionally set to true. Now, the options are properly set from chan_dahdi.conf. (closes issue #15508) Reported by: elguero Tested by: elguero ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	263df0044d	Merged revisions 206939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206939 \| dvossel \| 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) \| 20 lines Merged revisions 206938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206938 \| dvossel \| 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) \| 14 lines SIP incorrect From: header information when callpres is prohib Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	b66607d448	Blocked revisions 206877 via svnmerge ........ r206877 \| dvossel \| 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009) \| 6 lines TIMEOUT(absolute) returned negative value. (closes issue #15513) Reported by: ys ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	b400eb240e	Merged revisions 206873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206873 \| dvossel \| 2009-07-16 16:33:51 -0500 (Thu, 16 Jul 2009) \| 12 lines Merged revisions 206872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206872 \| dvossel \| 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) \| 6 lines error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206876 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	7304dedfaf	Merged revisions 206868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206868 \| dvossel \| 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009) \| 14 lines Merged revisions 206867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206867 \| dvossel \| 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) \| 8 lines avoid segfault caused by user error If the CALLERPRES() dialplan function is set to nothing, a segfault occurs. This is user error to begin with, but I'd rather see a cli warning message than have Asterisk crash on me. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206871 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5d94f8e6b9	Merged revisions 206808 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206808 \| tilghman \| 2009-07-16 11:51:05 -0500 (Thu, 16 Jul 2009) \| 13 lines Merged revisions 206807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206807 \| tilghman \| 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) \| 6 lines Fix a memory leak. (closes issue #15517) Reported by: adomjan Patches: func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	0faed3d459	Merged revisions 206768 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 \| dvossel \| 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) \| 8 lines Session timer were not activated if Supported header field in INVITE had both "timer" and other options. (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	a562b6afa9	Blocked revisions 206767 via svnmerge ........ r206767 \| jpeeler \| 2009-07-15 17:02:55 -0500 (Wed, 15 Jul 2009) \| 10 lines The dialing flag was mistakingly removed from sig_pri. This readds the proper setting of the flag and is really a continuation of r205731. The flag was being set properly in sig_analog, but use of the newly added set_dialing callback allowed for some simplification in chan_dahdi. (closes issue #15486) Reported by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	57f664c8f4	Merged revisions 206707 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206707 \| rmudgett \| 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) \| 33 lines Merged revisions 206706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206706 \| rmudgett \| 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) \| 26 lines Merged revision 206700 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f84624e23d	Merged revisions 206702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 \| dvossel \| 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) \| 10 lines callerid(num) is wrong when username is missing A domain only sip uri <sip:123.123.123.123> would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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