asterisk

Commit Graph

Author	SHA1	Message	Date
Michiel van Baak	7244366e7a	libxml2-dev is needed as well by default. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	126bf8eeb5	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/trunk@208813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b7cfe90404	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/trunk@208749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	742f0b90dd	Remove trailing whitespace. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	d79f6e17fa	Note that "reload" needs to be added back. I keep getting annoyed at having to type "module reload" to reload everything, so I'm adding a note that we need to add "reload" back. "module reload" doesn't really make sense as the command to reload everything, including the core. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	5d28d72d37	Don't log a warning for something that does not affect operation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	99025f09d7	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/trunk@208630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0b2b01a1fa	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/trunk@208593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	554c5e62d0	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/trunk@208588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	17e2d9fdbc	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/trunk@208548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	3ee2e7566f	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/trunk@208542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	347665503e	T.38 change note is not necessary in this branch git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	0a6e06c7ff	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/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	88f1d14766	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/trunk@208388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	dcd6227f6c	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/trunk@208383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bacf6ab51e	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/trunk@208314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	980db1601a	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/trunk@208267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	98b4bdc1b9	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/trunk@208263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	3843480b8f	Fix potential crash if p->owner is NULL. Problem was observed when a call-forwarding loop was accidentally configured. ABE-1906 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	b7928de6ac	Resolve compiler warning on mac. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208193 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	58699809a5	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/trunk@208155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d223e3636f	Merged revisions 208083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208083 \| tilghman \| 2009-07-22 15:23:53 -0500 (Wed, 22 Jul 2009) \| 4 lines Export symbols for functions included in our compatibility headers. (closes issue #15556) Reported by: smw1218 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	d009896670	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/trunk@208113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	4ff3f0058d	Clarify documentation on 'realtime update2' to show more than one condition. (closes issue #15357) Reported by: snuffy Patches: bug_fix_doc_update2.diff uploaded by snuffy (license 35) (slightly modified by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	299a9ff3fa	Remove trailing whitespace. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	33852cfaf6	Fix the crash in directed pickups. For real this time. A shallow pointer copy was causing an ast_party_connected_line structure to be freed multiple times, thus causing a crash. (closes issue #15441) Reported by: lmsteffan Patches: 15441.patch uploaded by mmichelson (license 60) Tested by: lmsteffan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	16328efb78	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/trunk@207950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5484d2f5d0	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/trunk@207946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	56c59985de	whitespace fix only git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	ced2554f60	Note that we use tabs instead of spaces for indentation. I'm surprised this was never actually in here... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	7466e00663	Fix my_is_off_hook to check rxbits only for FXS signaling git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	6ac23c3eca	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/trunk@207854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b1d9b989ee	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/trunk@207723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	96e4e31eeb	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/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	fe0de896f0	Blocked revisions 207573 via svnmerge ........ r207573 \| jpeeler \| 2009-07-20 18:23:18 -0500 (Mon, 20 Jul 2009) \| 10 lines 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/trunk@207599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	d040266a17	Okay, that didn't fix the crash. It didn't really do anything useful. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b276189912	Initialize connected line instance when doing a directed pickup. This helps to prevent a crash which may occur due to our freeing garbage due to a struct being uninitialized. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3f8059f87d	reg->username is parsed only once on sip reload The registration string can contain an expanded user portion of the form user@domain. This expanded user portion was stored in reg->username and parsed each time there is a registration refresh. Now, the domain portion of the user is parsed and stored separately in the regdomain field. (closes issue #14331) Reported by: Nick_Lewis Patches: chan_sip.c.domainparse3.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bec894cbe5	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/trunk@207424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	44301c95d2	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/trunk@207361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	bcff592839	Merged 207316 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r207316 \| rmudgett \| 2009-07-17 23:05:05 -0500 (Fri, 17 Jul 2009) \| 20 lines Fixed incoming calls being matched to MSNs without type-of-number prefix added. For an incoming ISDN call the dialed.number is incorrectly matched against the configured MSNs in misdn.conf. The numbers passed to the dialplan include the configured prefix for the dialed.number_type, whereas the check against the configured MSNs (to decide if the call is accepted at all), is executed without the configured prefix. e.g., dialed.number = 241168020, TON = national, configured national prefix is "0". (This is the TON which is used by ISDN providers in the Netherlands.) In chan_misdn.c:cb_events() in case EVENT_SETUP the call to misdn_cfg_is_msn_valid() uses the unnormalized number 241168020, but 57 lines later the call to read_config() adds the prefix, and the dialed.number is now 0241168020, which is then used in the dialplan. misdn_cfg_is_msn_valid() must use the normalized number, too. JIRA ABE-1912 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	98e4ab5716	Flag field in wrong position. Reported by "Hoggins!" on asterisk-dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	f9dab29054	Recorded merge of revisions 145293,158010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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/trunk@207285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	165ff1314b	Add flag here, too (as requested by jsmith) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	090066be3b	fixes an error in r203638 CEL commit (closes issue #15525) Reported by: elguero Patches: iax2-double-unlock.patch uploaded by elguero (license 37) 15525.diff uploaded by dvossel (license 671) Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	aa379bb741	Document the "flag" field in the voicemessages table. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	74de8256bd	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/trunk@207156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	496b509c42	Update some missing allowed options for overlapdial git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207095 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	06cace8c42	Blocked revisions 207092 via svnmerge ........ r207092 \| jpeeler \| 2009-07-17 14:13:27 -0500 (Fri, 17 Jul 2009) \| 11 lines Enhance configuration option for overlapdial allowing direction choice Previously overlap dialing could only be turned on or off for both incoming and outgoing calls. New parameters incoming, outgoing, and both have been added to allow further control. There is no change in default behavior with these new options and allows in band DTMF to be accepted in one direction if required. (closes issue #14471) Reported by: eboscani ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	8c4d0ace54	Blocked revisions 207033 via svnmerge ........ r207033 \| dvossel \| 2009-07-17 13:00:38 -0500 (Fri, 17 Jul 2009) \| 4 lines sip option flags handled incorrectly (issue #15376) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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