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2010-06-14Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.rmudgett1-0/+49
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
2010-06-07Extract sig_ss7 out of chan_dahdi.rmudgett1-7/+8
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
2010-06-07Moved AOC request code out of the middle of code parsing the dialed number.rmudgett1-15/+14
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268734 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-04Incoming overlap dialing no longer works after sig_pri extraction.rmudgett1-8/+1
The problem would manifest itself if your dialplan matching could accept more digits to match than were actually dialed. The time out waiting for overlap digits disconnected the call instead of matching any accumulated digits to the dialplan. Accidental conversion of a break out of loop as a break out of switch. (closes issue #17401) Reported by: avalentin Patches: issue17401_digit_timeout.patch uploaded by rmudgett (license 664) Tested by: avalentin, rmudgett git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267928 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Add ETSI Message Waiting Indication (MWI) support.rmudgett1-1/+199
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 Malicious Call ID support.rmudgett1-0/+110
Add the ability to report malicious callers as an AMI event in the call event class. Relevant specification: EN 300 180 Review: https://reviewboard.asterisk.org/r/576/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267350 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Call Waiting support.rmudgett1-112/+458
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.rmudgett1-427/+916
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 Advice Of Charge (AOC) event reporting.rmudgett1-0/+683
This feature generates AMI events in the new aoc event class from the events passed up by libpri. Review: https://reviewboard.asterisk.org/r/537/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267008 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Explicit Call Transfer (ECT) support.rmudgett1-74/+123
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
2010-05-11Dialing an invalid extension causes incomplete hangup sequence.rmudgett1-6/+36
Revision -r1489 of the libpri 1.4 branch corrected a deviation from Q.931 Section 5.3.2. However, this resulted in an unexpected behaviour change to the upper layer (Asterisk). This change uses pri_hangup_fix_enable() to follow Q.931 Section 5.3.2 call hangup better if the version of libpri supports it. (issue #17104) Reported by: shawkris Tested by: rmudgett git-svn-id: http://svn.digium.com/svn/asterisk/trunk@262569 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-07Fix deadlock in sig_pri when hanging up.jpeeler1-10/+10
The pri_dchannel thread currently violates locking order by locking the private and then attempting to queue a frame, which needs to lock the channel. Queueing a frame is unneccesary though and is actually a regression since sig_pri. All the places that currently use ast_softhangup_nolock now will just set the softhangup value directly as before. (closes issue #17216) Reported by: lmsteffan Patches: bug17216.patch uploaded by jpeeler (license 325) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261866 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Some code optimizations.rmudgett1-80/+88
* Made more places use pri_queue_control() instead of pri_queue_frame() and a local frame variable. * Made pri_queue_frame() use sig_pri_lock_owner(). pri_queue_frame() no longer releases the libpri access lock unless it is required. * Made the pri_queue_frame() and pri_queue_control() parameter list similar to sig_pri_lock_owner(). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261822 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04The inalarm flag is not passed up from the sig_analog and sig_pri submodules.rmudgett1-12/+20
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
2010-04-30Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7, and MFCR2 users.rmudgett1-13/+13
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
2010-04-09Merge Call completion support into trunk.mmichelson1-40/+1512
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
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.rmudgett1-9/+4
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
2010-03-10Simplified dahdi_request() channel selection failed reason/cause code.rmudgett1-10/+5
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.rmudgett1-72/+51
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
2010-02-27overlap receiving: automatically send CALL PROCEEDING when dialplan startsalecdavis1-0/+12
Following Q.931 5.2.4 When the user has determined that sufficient call information has been received the user shall stop T302 and send CALL PROCEEDING to the network. Previously timeouts were possible if the dialplan took a long time to issue any response back to the network. Verified that our local TELCO also does the same. (issue #16789) Reported by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249320 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-20Update CDR variables as pbx startsalecdavis1-2/+0
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call, by executing ast_cdr_update() early in __ast_pbx run. Reverts sig_pri changes in trunk that are specific to isdn technology only. (closes issue #16638) Reported by: alecdavis Patches: cdr_update.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241416 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-19Update CDR variables before pbx starts (overlap dial)alecdavis1-0/+2
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call. (issue #16638) Reported by: alecdavis Patches: cdr_update.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241187 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-18Update CDR variables before pbx startsalecdavis1-0/+2
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call. (closes issue #16638) Reported by: alecdavis Patches: cdr_update.diff.txt uploaded by alecdavis (license 585) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241097 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Fix using the wrong pointer type in do_idle_thread().rmudgett1-4/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@238527 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-05Removed unused parameters from analog_available() and sig_pri_available().rmudgett1-1/+1
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 ↵kpfleming1-26/+26
feature release git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235904 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-24Fix problem on digital channels due to digital flag not getting set jpeeler1-6/+18
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
2009-11-06Created standard location to add options to chan_dahdi for ISDN dialing.rmudgett1-71/+56
Dial(DAHDI/g1[/extension[/options]]) Current options: K(<keypad_digits>) R Reverse charging indication (Collect calls) The earlier Dial(DAHDI/g1[/K<keypad_digits>][/extension] format was variable and did not allow for the easy addition of more options. The earlier 'C' prefix character for reverse charge indiation would conflict with the a-d DTMF digits if ISDN uses them. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228691 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Expand codec bitfield from 32 bits to 64 bits.tilghman1-13/+13
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02DAHDI ISDN channel names will not allow device state to work. (Interim ↵rmudgett1-0/+119
solution.) Since ISDN works like SIP and not analog ports in regard to devices, the device state based on the ISDN channel number could not work. This has not been an issue until the advent of PTMP NT mode. Previously, ISDN lines were used as trunks and did not have to keep track of specific devices. As an interim solution until device states are properly implemented, the channel name is being changed to the following format to use the generic device state support: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> Dialplan hints would thus be: exten => xxx,hint,DAHDI/i2/5551212 This will work with the following restrictions: * The number of devices/phones cannot exceed the number of B channels. (i.e., BRI has 2) * Each device/phone can only have one number. No shared MSN's. * The phones/devices probably should not use subaddressing. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226882 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-30Cleanup some flags on DAHDI PRI channel hangup.rmudgett1-1/+8
* Cleanup some flags on DAHDI PRI channel hangup. (sig_pri split) * Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls. * Remove some unused flags since sig_pri was split. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226648 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-23Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.rmudgett1-103/+644
* 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
2009-10-22Search for the subaddress only within the extension section of the dial string.rmudgett1-16/+17
Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension]) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225446 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-22Add support for calling and called subaddress. Partial support for COLP ↵rmudgett1-1/+292
subaddress. The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing "desk to desk" between offices each with an asterisk box over the ISDN should then be possible, without a whole load of DDI numbers required. (closes issue #15604) Reported by: alecdavis Patches: asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585) Some minor modificatons were made. Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/405/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225357 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Make PRI_SUBCMD_xxx handling subaddress friendly.rmudgett1-68/+75
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224930 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Add a callback to sig_pri which is called when sig_pri is going to queue a ↵file1-0/+4
control frame on a channel. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224491 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-17Merged revisions 224330 via svnmerge from jpeeler1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224331 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-16Merged revisions 224260 via svnmerge from rmudgett1-4/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (issue #14292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224261 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-28Miscellaneous minor changes.rmudgett1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220792 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-28Locking issues dealing with service_lock.rmudgett1-15/+10
* Removed unneeded and uninitialized service_lock. * Fixed potential locking imbalance in pri_dchannel():PRI_EVENT_RESTART. * Fixed verbose message typo in pri_dchannel():PRI_EVENT_RESTART. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220672 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-25Reduce indentation in sig_pri_available().rmudgett1-18/+16
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220543 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-10Cleanup approach in 217804 and don't reach inside the sig_pvt.jpeeler1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217987 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-08Fix memory leak of sig_xxx private structures.rmudgett1-0/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217332 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-08Remove duplicate entry in the sig_pri_pri private pointer array.rmudgett1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217236 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03Lets try not to use C++ keywords for variable names.rmudgett1-14/+16
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@216186 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/+42
ISDN PTMP CPE spans. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@215757 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-28Move discardremoteholdretrieval test so it applies only to the specific ↵rmudgett1-7/+13
notification indicator values. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@214654 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-21Update configure script for libpri COLP feature dependency requirements.rmudgett1-6/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213748 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-18Add COLP support to chan_dahdi/sig_pri.rmudgett1-60/+543
Add Connected Line Presentation (COLP) support to chan_dahdi/libpri as an addition to issue 8824. This is the chan_dahdi/sig_pri portion. COLP support is now available for any switch for which libpri supports COLP (currently ETSI PTP, ETSI PTMP, and Q.SIG) with this patch. (closes issue #14068) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/340/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213007 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-15 09:04:51 +0000