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2011-02-02Replacing doc/* with wiki linkslathama1-2/+2
Adding links to http(s)://wiki.asterisk.org git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305799 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-27Merged from revision 304341rmudgett1-16/+43
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed, 26 Jan 2011) | 7 lines Add connected line chan_dahdi.conf pricpndialplan option. * Added from_channel value to prilocaldialplan option. JIRA ABE-2731 JIRA SWP-2842 .......... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@304385 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Optional HOLD/RETRIEVE signaling for PTMP TE when the bridge goes on and off ↵rmudgett1-4/+11
hold. Added the moh_signaling option to specify what to do when the channel's bridged peer puts the ISDN channel on and off of hold. Implemented as a FSM to control libpri ISDN signaling when the bridged peer places the channel on and off of hold with the AST_CONTROL_HOLD and AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691 Review: https://reviewboard.asterisk.org/r/1063/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@300212 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-22Merged revisions 295869 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295869 | seanbright | 2010-11-22 15:03:49 -0500 (Mon, 22 Nov 2010) | 9 lines Merged revisions 295868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, 22 Nov 2010) | 2 lines Change some documentation to suggest dahdi_monitor instead of ztmonitor. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@295870 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13Merged revisions 286426 via svnmerge from rmudgett1-2/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286426 | rmudgett | 2010-09-13 10:52:14 -0500 (Mon, 13 Sep 2010) | 1 line Update chan_dahdi.conf.sample to reflect new libpri T309 default value. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@286427 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-22Support FXS module Polarity Reversal on remote party Answer and Hangup alecdavis1-5/+13
FXS lines normally connect to a telephone. However, when FXS lines are routed to an external PBX or Key System to act as "external" or "CO" lines, it is extremely difficult, if not impossible for the external PBX to know when the call has been disconnected without receiving a polarity reversal on the line. Now using answeronpolarityswitch and hanguponpolarityswitch keywords that previously were used only for FXO ports, now applies like functionality for an FXS port, but from the connected equipment's point of view. (closes issue #17318) Reported by: armeniki Patches: fxs_linepolarity.diff5.txt uploaded by alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/797/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278809 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.rmudgett1-0/+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
2010-06-09dial by name in chan_dahditzafrir1-1/+26
* 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
2010-06-03Add ETSI Message Waiting Indication (MWI) support.rmudgett1-0/+6
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.rmudgett1-0/+11
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/+18
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.rmudgett1-0/+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
2010-04-27Merged revisions 259270 via svnmerge from rmudgett1-4/+4
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
2010-04-09Merge Call completion support into trunk.mmichelson1-4/+52
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
2010-03-03Add new config option to control AMI alarm event reporting in chan_dahdi.jpeeler1-0/+8
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 configurationdvossel1-0/+7
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
2009-10-23Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.rmudgett1-0/+6
* Added handling of received HOLD/RETRIEVE messages and the optional ability to transfer a held call on disconnect similar to an analog phone. * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP. Will reroute/deflect an outgoing call when receive the message. Can use the DAHDISendCallreroutingFacility to send the message for the supported switches. * Added ability to send/receive keypad digits in the SETUP message. Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension]) Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)} * Added support for BRI PTMP NT mode. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225692 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Add dynamic range compression support for analog channels.mnicholson1-0/+12
(closes issue AST-29) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224637 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-15chan_dahdi.conf.sample changes for DTMF CID detect dbailey1-0/+11
Explains new options for detecting DTMF CID on fxo lines (issue #9096) Reported by: fleed Patches: chan_dahid_sample_config.patch uploaded by sum (license 766) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224144 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-02Made chan_dahdi able to ignore incoming calls that are not in a MSN list for ↵rmudgett1-0/+8
ISDN PTMP CPE spans. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@215757 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-26Minor punctuation change.rmudgett1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@214272 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-17Update some missing allowed options for overlapdialjpeeler1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207095 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-14fix a typo in sample config file for option changejpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206603 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-02Support setting and receiving Reverse Charging Indication over ISDN PRI.seanbright1-0/+5
This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse Charging Indication in LibPRI. This patch adds the ability to specify RCI on the outbound leg of a PRI call from within Asterisk, by prefixing the dialed number with a capital 'C' like: ...,Dial(DAHDI/g1/C4445556666) And to read it off an inbound channel: exten => s,1,Set(RCI=${CHANNEL(reversecharge)}) Thanks again to rmudgett for the thorough review. (closes issue #13760) Reported by: mrgabu Review: https://reviewboard.asterisk.org/r/303/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204749 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-25Remove some unnecessary code and update sample config file with respect to ↵jpeeler1-9/+1
GR-303. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203402 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-16keep backwards compatible chan_dahdi with older openr2 versions by not using ↵moy1-3/+3
the new skip category feature unless supported git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200799 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-14added openr2 to menuselect-deps.in, recent commit in menuselect made me ↵moy1-6/+20
realize this was never done but was working anyways also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200477 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.seanbright1-65/+65
Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197535 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.seanbright1-14/+14
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197528 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Fix references to /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf inseanbright1-4/+5
the sample configuration files. (closes issue #15207) Reported by: seandarcy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197089 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-14Add service maintenance message supportjpeeler1-0/+3
This is the companion commit to libpri r732. Service messages are now supported for switch types 4ess/5ess. A new option service_message_support has been added to chan_dahdi.conf and is noted in the sample config file. The service message support is turned off by default. The current implementation relies on AstDB to keep track of channel state, which allows the statuses to be preserved across Asterisk restarts. Below is a description of the storage format. The state and reason for the service state are in the form <state>:<reason>, where: <state> ::= { 'O' } // 'O' – Out Of Service <reason> ::= { '0' | '1' | '2' | '3' }, where: '0' – No reason (backwards compatibility) '1' – NEAR END '2' – FAR END '3' – both NEAR and FAR END The new CLI commands to handle channel service state are: pri service disable channel <chan> pri service enable channel <chan> Many people contributed to the development of this functionality. Because I entered at the very end I do not know the exact history. Special thanks to all who moved the bug forward one way or another: cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7, tilghman, lmadsen, and especially dhubbard (he answered lots of my questions and did a large portion of the work) (closes issue #3450) Reported by: cmaj git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188342 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-16Add MFC/R2 support for chan_dahdi.russell1-0/+149
This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182355 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Add dynamic fax buffer configuration option to chan_dahdi.confdhubbard1-0/+9
When the 'faxdetect' configuration option is used, one may also want to use the 'faxbuffers' configuration option in chan_dahdi.conf. This option will dynamically use the configured 'faxbuffers' buffer policy on a channel for the life of the call following the detection of fax tones. The faxbuffers buffer policy will be reverted during call teardown. An example use of 'faxbuffers' is below. This example would switch to using 6 buffers with a full buffer policy. faxbuffers=>6,full git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175411 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29channels/chan_dahdi.crmudgett1-3/+5
* Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172400 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-19Add enhanced MWI generation to take advantage of new dahdi line reversal MWI ↵dbailey1-0/+11
ability. (closes issue #14104) Reported by: alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey (license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, dbailey git-svn-id: http://svn.digium.com/svn/asterisk/trunk@169277 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-17Add discriminator for when ring pulse alert signal is used to preface MWI spillsdbailey1-6/+14
This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up git-svn-id: http://svn.digium.com/svn/asterisk/trunk@169153 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-25Add an option, waitfordialtone, for UK analog lines which do not end a calltilghman1-0/+7
until the originating line hangs up. (closes issue #12382) Reported by: one47 Patches: zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463) Tested by: fleed git-svn-id: http://svn.digium.com/svn/asterisk/trunk@159317 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-09Fix this as well. Pointed out by tzafrir.seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155671 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-09Fix some spelling errors, and convert tabs to spaces.seanbright1-38/+33
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155637 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-17Merge in patch for #13454. Includes CallRereouting dialplan application, ↵mattf1-0/+14
option for discard of remote hold messages, and using the alternate logical channel mapping in Q.SIG instead of the default physical channel mapping. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150640 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-04Merged revisions 135536 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) | 2 lines fix a config sample typo ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135537 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-04Merged revisions 135473 via svnmerge from russell1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008) | 2 lines Add a minor clarification to the documentation of mohinterpret and mohsuggest ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135474 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-30Move implementation of an attended-transfer-complete sound from one channeltilghman1-1/+5
driver into a common place for multiple channel drivers. (closes issue #13152) Reported by: caio1982 Patches: atxfer_complete_sound3.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-28remove remaining Zaptel references in various placeskpfleming1-7/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134086 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-22Merged revisions 132641 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul 2008) | 2 lines use renamed libpri API call for controlling this feature (was improperly named before) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132643 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11Merged revisions 130039 via svnmerge from kpfleming1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today (related to issue #13042) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130040 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Add option to wait to be able to explicitly send ACM via the Proceeding() ↵mattf1-0/+11
application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128122 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-30rename zapata.conf.sample to chan_dahdi.conf.samplejpeeler1-0/+981
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126675 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-30 21:09:44 +0000