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2010-11-24Merged revisions 296166 via svnmerge from rmudgett1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296166 | rmudgett | 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) | 50 lines Merged revisions 296165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip. The FXS connected phone has to have CW/CID support to fail, as it will send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal phone with no CID never fails. Also the SIP phone does not hear MOH when the CW call is answered. The DTMF end frame is suppressed when the phone acknowledges the CW signal for CID. The problem is the DTMF begin frame needs to be suppressed as well. The DTMF begin frame is causing SIP to start sending the DTMF RTP frames. Since the DTMF end frame is suppressed, SIP will not stop sending those DTMF RTP packets. * Suppress the DTMF begin and end frames when the channel driver is looking for DTMF digits. * Fixed a couple issues caused by not cleaning up the CID spill if you answer the CW call while it is sending the CID spill. * Fixed not sending CW/CID spill to the phone when the call is natively bridged. (Fixed by not using native bridge if CW/CID is possible.) * Suppress received audio when sending CW/CID spills. The other parties involved do not need to hear the CW/CID spills and may be confused if the CW call is for them. (closes issue #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett NOTE: * v1.4 does not have the main problem fixed by suppressing the DTMF start frames. The other three items fixed are relevant. * If you really must restore native bridging between analog ports, you need to disable CW/CID either by configuring chan_dahdi.conf callwaitingcallerid=no or dialing *70 before dialing the number to temporarily disable CW. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296167 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Merged revisions 294822 via svnmerge from rmudgett1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) | 11 lines Asterisk is getting a "No D-channels available!" warning message every 4 seconds. Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294823 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Merged revisions 285710 via svnmerge from bbryant1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Rename sig_pri_pri to sig_pri_span. More descriptive of concept.rmudgett1-16/+16
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278942 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.rmudgett1-0/+12
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
2010-06-08Fix 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.rmudgett1-1/+1
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/+42
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.rmudgett1-5/+41
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/+17
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/+5
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
2010-05-04The inalarm flag is not passed up from the sig_analog and sig_pri submodules.rmudgett1-2/+3
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.rmudgett1-10/+10
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
2010-04-09Remove PRI CCSS BUGBUG message and update configure script.rmudgett1-4/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256569 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Merge Call completion support into trunk.mmichelson1-0/+102
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
2010-03-11Minor tweaks and comment updates to chan_dahdi.rmudgett1-2/+9
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.rmudgett1-1/+1
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.rmudgett1-6/+10
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
2010-01-05Removed 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
2009-11-24Fix problem on digital channels due to digital flag not getting set jpeeler1-1/+2
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
2009-11-02DAHDI ISDN channel names will not allow device state to work. (Interim ↵rmudgett1-0/+2
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
2009-10-23Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.rmudgett1-0/+21
* 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 a callback to sig_pri which is called when sig_pri is going to queue a ↵file1-0/+1
control frame on a channel. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224491 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Prevent deadlock if chan_dahdi attempts to change PRI channel names.rmudgett1-0/+1
The PRI channels can no longer change the channel name if a different B channel is selected during call negotiation. To prevent using the channel name to infer what B channel a call is using and to avoid name collisions, the channel name format is changed. The new channel naming for PRI channels is: DAHDI/ISDN-<span>-<sequence-number> git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221701 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-28Locking issues dealing with service_lock.rmudgett1-1/+0
* 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
2009-09-10Cleanup approach in 217804 and don't reach inside the sig_pvt.jpeeler1-0/+2
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/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217332 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/+1
ISDN PTMP CPE spans. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@215757 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-18Add COLP support to chan_dahdi/sig_pri.rmudgett1-2/+0
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
2009-08-10Restoring some code to sig_pri. Not sure if it is really needed.rmudgett1-0/+1
Putting some DSP code back into sig_pri that was removed by the chan_dahdi/sig_pri reorganization. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211392 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-05More changes from chan_dahdi that did not make it into sig_pri.rmudgett1-0/+4
* Q.SIG channel mapping option. * discardremoteholdretrieval option. * libPRI debug defines. * pri_set_overlapdial() now set correctly. * pthread creation of pri_ss_thread now matches. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210696 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-04Fix CALLERID() values for sig_pri on incoming calls.rmudgett1-7/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210387 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-03Trim trailing whitespace.rmudgett1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210094 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-30Add missing ifdef-s for service maintenance message functionalityjpeeler1-0/+2
(closes issue #15614) Reported by: fabled git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209619 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-30Fixes numerous spelling errors. Patch submitted by alecdavis.dbrooks1-1/+1
(closes issue #15595) Reported by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209554 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-23Fix sending of interface identifier unconditionally in sig_prijpeeler1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@208267 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-15The dialing flag was mistakingly removed from sig_pri.jpeeler1-0/+1
This readds the proper setting of the flag and is really a continuation of r205731. The flag was being set properly in sig_analog, but use of the newly added set_dialing callback allowed for some simplification in chan_dahdi. (closes issue #15486) Reported by: rmudgett git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206767 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-03Add a configure check for Reverse Charging Indication support in LibPRI.seanbright1-0/+2
Also go back and wrap all of the places that use the specific reverse charge APIs with preprocessor conditionals. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204919 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-02Support setting and receiving Reverse Charging Indication over ISDN PRI.seanbright1-0/+1
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
2009-06-26Merge the new Channel Event Logging (CEL) subsystem.russell1-2/+2
CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-25New signaling module to handle PRI/BRI operations in chan_dahdijpeeler1-0/+257
This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into sig_pri.c. Functionality in theory should not change (mostly). A few trivial changes were made in sig_analog with verbose messages and commenting. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203304 f38db490-d61c-443f-a65b-d21fe96a405b
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