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2010-06-14Extract sig_ss7_init_linkset() to sig_ss7.rmudgett1-16/+5
Also found a place where sig_pri_init_pri() was inlined and called it instead. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270298 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.rmudgett1-1/+16
Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn. Review: https://reviewboard.asterisk.org/r/696/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270219 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-10Attempt to fix a FreeBSD build error by including sys/stat.h.russell1-0/+1
http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269602 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Add missing API function to sig_ss7: sig_ss7_fixup().rmudgett1-7/+14
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269308 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Eliminate deadlock potential in dahdi_fixup().rmudgett1-3/+4
Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are in a potentially inconsistent state is a potentially bad thing in principle. However, calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269307 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09dial by name in chan_dahditzafrir1-10/+122
* chan_dahdi supports dialing configuring and dialing by device file name. DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise it may appear in chan_dahdi.conf as 'channel => span-name!local!1'. * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean. False by default. If set, chan_dahdi will ignore failed 'channel' entries. Handy for the above name-based syntax as it does not depend on initialization order. * have my_pri_make_cc_dialstring() only manupulate dial-strings of group (gGrR) dialing, which make it lsightly more complicated. https://reviewboard.asterisk.org/r/535/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269238 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Fix some doxygen warnings.lmadsen1-1/+1
(closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268969 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Extract sig_ss7 out of chan_dahdi.rmudgett1-1571/+749
Extract the SS7 specific code out of chan_dahdi like what was done to ISDN/PRI and analog signaling. The new SS7 structures were modeled on sig_pri. The changes to sig_pri are an enhancement and a bug fix made possible because SS7 was extracted. 1) The sig_pri TRANSFERCAPABILITY channel variable should have been set unconditionally in sig_pri_new_ast_channel(). 2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of SS7 extraction. 3) Module ref count error in dahdi_new() if startpbx failed to start the PBX for some reason. Review: https://reviewboard.asterisk.org/r/661/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268774 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Add ETSI Message Waiting Indication (MWI) support.rmudgett1-0/+11
Add the ability to report waiting messages to ISDN endpoints (phones). Relevant specification: EN 300 650 and EN 300 745 Review: https://reviewboard.asterisk.org/r/599/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267399 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Call Waiting support.rmudgett1-36/+240
Add the ability to announce a call to an endpoint when there are no B channels available. A call waiting call is a SETUP message with no B channel selected. Relevant specification: EN 300 056, EN 300 057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the "no_media_path" option. * Returns "0" if there is a B channel associated with the call. * Returns "1" if no B channel is associated with the call. The call is either on hold or is a call waiting call. If you are going to allow incoming call waiting calls then you need to use CHANNEL(no_media_path) do determine if you must drop a call to accept the new call. Review: https://reviewboard.asterisk.org/r/568/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267261 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Generic Advice of Charge.rmudgett1-0/+19
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Explicit Call Transfer (ECT) support.rmudgett1-1/+2
Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages to eliminate tromboned calls. Note: Asterisk already supported initiating the transfer of calls to eliminate tromboned calls to libpri so there was nothing to do for the asterisk portion. Review: https://reviewboard.asterisk.org/r/520/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266926 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17Enhancements to connected line and redirecting work.mmichelson1-0/+10
From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-13Fix inverted logic in cli command: ss7 set debug on/offrmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263069 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-12Don't crash when destroying chan_dahdi pseudo channels.rmudgett1-1/+10
Must do a deep copy of the cc_params in duplicate_pseudo(). Otherwise, when the duplicate pseudo channel is destroyed, it frees the original pseudo channel cc_params. The original pseudo channel is then left with a dangling pointer for when the next duplicated pseudo channel is created. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@262743 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06When failing to configure, don't destroy 'cfg' twicetzafrir1-1/+1
Fixes a crash when some config section had an incorrect channel config. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261451 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05Avoid a crash on SS7 channels.rmudgett1-2/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261405 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04The inalarm flag is not passed up from the sig_analog and sig_pri submodules.rmudgett1-14/+53
The CLI "dahdi show channel" command was not correctly reporting the InAlarm status. The inalarm flag is now consistently passed between chan_dahdi and submodules. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261007 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Merged revisions 260434 via svnmerge from jpeeler1-0/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) | 11 lines Ensure channel state is not incorrectly set in the case of a very early answer. The needringing bit was being read in dahdi_read after answering thereby setting the state to ringing from up. This clears needringing upon answering so that is no longer possible. (closes issue #17067) Reported by: tzafrir Patches: needringing.diff uploaded by tzafrir (license 46) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260437 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7, and MFCR2 users.rmudgett1-31/+33
Created SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS SIG_MFCR2_MAX_CHANNELS Also fixed the declaration of pollers[] in mfcr2_monitor(). It was dimensioned to the number of bytes in struct dahdi_mfcr2.pvts[] and not to the same dimension of the struct dahdi_mfcr2.pvts[]. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260435 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-29Merged revisions 260195 via svnmerge from rmudgett1-12/+22
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) | 26 lines DTMF CallerID detection problems. The code handling DTMF CallerID drops digits on long CallerID numbers and may timeout waiting for the first ring with shorter numbers. The DTMF emulation mode was not turned off when processing DTMF CallerID. When the emulation code gets behind in processing the DTMF digits it can skip a digit. For shorter numbers, the timeout may have been too short. I increased it from 2 seconds to 4 seconds. Four seconds is a typical time between rings for many countries. (closes issue #16460) Reported by: sum Patches: issue16460.patch uploaded by rmudgett (license 664) issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA AST-334 JIRA SWP-901 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260231 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Merged revisions 259531 via svnmerge from rmudgett1-6/+24
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success" Changed the warning to "Failed to decode CallerID on channel 'name'". The message before it is likely more specific about why the CallerID decode failed. SWP-501 AST-283 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Merged revisions 259270 via svnmerge from rmudgett1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259307 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Re-fix dahdi_request() iflist locking since CCSS merged.rmudgett1-2/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259229 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-16Enable PRI SERVICE message support in chan_dahdi for the 'national' switchtypedhubbard1-4/+7
Revision 1072 of libpri added SERVICE message support for the 'national' switchtype. The attached patch enables the use of 'pri service' CLI commands on dahdi channels that are configured for the 'national' switchtype. (closes issue #17142) Reported by: dhubbard Patches: dw-ni2.patch uploaded by dhubbard (license 733) Tested by: elguero, dhubbard Review: https://reviewboard.asterisk.org/r/612/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257642 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Merge Call completion support into trunk.mmichelson1-98/+817
From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256528 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-06Merged revisions 256225 via svnmerge from rmudgett1-18/+24
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock. SWP-1231 ABE-2163 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256265 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.rmudgett1-1/+1
SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256104 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25remove unneeded explicit channel in dahdi ioctlstzafrir1-17/+13
This patch removes some cases where the channel number for an ioctl was passed as a member in a struct rather then through the file descriptor. The gain setting functions passed around a channel which is always 0, and thus this parameter is simply dropped. Review: https://reviewboard.asterisk.org/r/584/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254406 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Use SHRT_MAX instead of MAXSHORT.russell1-3/+2
These changes fix build issues I had with this module on FreeBSD. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253536 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-12add missing mfcr2_skip_category settingmoy1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252088 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-12Doxegen this chan_dahdi lock.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251946 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-11Minor tweaks and comment updates to chan_dahdi.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251821 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-10Simplified dahdi_request() channel selection failed reason/cause code.rmudgett1-12/+13
Also avoid potential crash because cause could be NULL. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251585 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-10Reduce the amount of database access for HAVE_PRI_SERVICE_MESSAGES.rmudgett1-47/+52
Rework HAVE_PRI_SERVICE_MESSAGES to not use the active values directly from the database. Database access is likely expensive. Database access now only happens on initialization, destruction, and when the B channel is taken in or out of service. This change is not related to call waiting but it would cause the search for a call waiting interface to be very expensive and slow down D channel message servicing. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Removed cdrflags from ast_channel structure.rmudgett1-3/+0
Only chan_dahdi set a value in cdrflags. Everyone else just copied it around the system. Noone cared about any value it may have contained. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250565 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Merged revisions 250480 via svnmerge from jpeeler1-0/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) | 15 lines Make sure to clear red alarm after polarity reversal. From the issue: The automatic overnight line tests (or manual ones) used on UK (BT) lines causes a red alarm on a dahdi / TDM400P connected channel. This is because the line uses voltage tests (battery loss) and polarity reversal. The polarity reversal causes chan_dahdi to initiate v23 CallerID processing but during this the event DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared. (closes issue #14163) Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250481 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Add new config option to control AMI alarm event reporting in chan_dahdi.jpeeler1-17/+80
New config parameter "reportalarms" added in chan_dahdi.conf which supports the following possible values: "channels": report each channel alarms (current behavior, default for backward compatibility) "spans": report an "SpanAlarm" event when the span of any configured channel is alarmed "all": report channel and span alarms (aggregated behavior) "none": do not report any alarms (closes issue #16709) Reported by: nahuelgreco Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco (license 162) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250392 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02fixes adaptive jitterbuffer configurationdvossel1-1/+2
When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249893 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02fix asterisk setting of pritimers from chan_dahdi.confalecdavis1-4/+5
regression since sig_pri split. (issue #16909) Reported by: alecdavis Patches: pritimer.asterisk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249759 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19mfcr2 issue 0016844 - Fix portability bit fields and make ↵moy1-10/+10
mfcr2_immediate_accept work again, reported and patched by korihor git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248003 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-15Restore triedtopribridge flag code removed in -r211197.rmudgett1-1/+3
Ooops. Failed to note that we were inside a for loop and pri_channel_bridge() needs to be executed only once. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246669 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-03The chanvar= setting should inherit the entire list of variables, not just ↵tilghman1-1/+7
the first one. (closes issue #16359) Reported by: raarts Patches: dahdi-setvars.diff uploaded by raarts (license 937) Tested by: raarts git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244505 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-18Extend max call limit duration from 24.8 days to 292+ million years.jpeeler1-2/+2
If the limit was set past MAX_INT upon answering, the call was immediately hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup). The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been changed to return an int64_t to prevent overflow. Also the reporter suggested adding a message indicating the reason for the call hanging up. Given that the new limit is so much higher, the message (which would only really be useful in the overflow scenario) has been made a debug message only. (closes issue #16006) Reported by: viraptor git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241143 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-05Removed unused parameters from analog_available() and sig_pri_available().rmudgett1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237804 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-21Change all refererences to 1.6.3 to be 1.8, since that will be the next ↵kpfleming1-10/+10
feature release git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235904 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-17Fix call forwarding for analog phones.jpeeler1-1/+3
(closes issue #16440) Reported by: mmichelson git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235382 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Merged revisions 232090 via svnmerge from jpeeler1-1/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines Do not modify the gain settings on data calls. (The digital flag actually represents a data call.) (closes issue #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232091 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-01More 32->64 bit codec conversions.tilghman1-2/+2
In the process of swapping ULAW to a place in the extended codec space, we found several unhandled cases, where a 32-bit integer was still being used to handle a codec field. Most of these have been fixed with this commit, although there is at least one case (codec_dahdi) which depends upon outside headers to be altered before a conversion can be made. (Fixes AST-278, SWP-459) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231850 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-24Fix problem on digital channels due to digital flag not getting set jpeeler1-19/+28
Changed areas in sig_pri to set the digital flag using a callback that will also set the corresponding flag in chan_dahdi. Modified dahdi_request slightly so that if a bearer is marked as digital, that information is available when creating the new channel. (closes issue #16151) Reported by: alecdavis Patch based on bug_16151.diff.txt uploaded by alecdavis (license 585) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231058 f38db490-d61c-443f-a65b-d21fe96a405b
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