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2012-07-06Remove unnecessary generation of informational cause framesKinsey Moore
It is not necessary to generate information cause code frames on every protocol event that occurs. This removes all the instances where the frame was not conveying a cause code and was instead just conveying a protocol-specific message. This also corrects the generation of the message associated with disconnects for MFC/R2 to use the MFC/R2 specific text for the disconnect cause. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-26Unique Call ID logging Phases III and IVJonathan Rose
Adds call ID logging changes to specific channel drivers that weren't handled handled in phase II of Call ID Logging. Also covers logging for threads for threads created by systems that may be involved with many different calls. Extra special thanks to Richard for rigorous review of chan_dahdi and its various signalling modules. review: https://reviewboard.asterisk.org/r/1927/ review: https://reviewboard.asterisk.org/r/1950/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22Add HANGUPCAUSE hash support for analog and PRI DAHDI subtechsKinsey Moore
This is part of the DAHDI support for the Asterisk 11 "Who Hung Up?" project and covers the implementation for the technologies implemented in sig_analog.c and sig_pri.c. Tested on a local machine to verify protocol and cause information is available. Review: https://reviewboard.asterisk.org/r/1953/ (issue SWP-4222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15Multiple revisions 369001-369002Kevin P. Fleming
........ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines Add support-level indications to many more source files. Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines Add a script to enable finding source files without support-levels defined. ........ Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-30Fix deadlock when executing CLI "pri show channels" and "ss7 show channels" ↵Richard Mudgett
commands. * Fix sig_pri_lock_owner() to avoid deadlock properly. * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid deadlock properly. * Code ss7_grab() better. (closes issue ASTERISK-19854) Reported by: Jaxon Patches: jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded by rmudgett (Modified to do the same thing to sig_ss7) Tested by: Jaxon ........ Merged revisions 367976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367978 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before ↵Richard Mudgett
disconnecting the call. Some switches may not handle the call-deflection/call-rerouting message if the call is disconnected too soon after being sent. Asteisk was not waiting for any reply before disconnecting the call. * Added a 5 second delay before disconnecting the call to wait for a potential response if the peer does not disconnect first. (closes issue ASTERISK-19708) Reported by: mehdi Shirazi Patches: jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363734 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25Clear ISDN channel resetting state if the peer continues to use it.Richard Mudgett
Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in response to a RESTART request. * Made the second SETUP received after sending a RESTART request clear the channel resetting state as if the peer had sent the expected RESTART ACKNOWLEDGE before continuing to process the SETUP. The peer may not be sending the expected RESTART ACKNOWLEDGE. (issue ASTERISK-19608) (issue AST-844) (issue AST-815) Patches: jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified) ........ Merged revisions 363687 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 363688 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20Add original party id and reason support.Richard Mudgett
ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who was the original redirecting party of a call. * Added support for the original redirecting party and reason to the REDIRECTING function and the system core as well as to the stubbed locations in sig_pri.c. Review: https://reviewboard.asterisk.org/r/1829/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-18Add ability to ignore layer 1 alarms for BRI PTMP lines.Richard Mudgett
Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362429 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-24Make number not available presentation also set screening to network provided.Richard Mudgett
Q.951 indicates that when the presentation indicator is "Number not available due to interworking" for a number then the screening indicator field should be "Network provided". * Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE when the presentation is "Number not available due to interworking". This fix makes Asterisk consistent and it also makes it consistent with earlier branches as far as this presentation value is concerned. * Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions handle the "Number not available due to interworking" case better in sig_pri.c. This change is possible because the minimum required libpri version (v1.4.11) has the necessary defines in libpri.h. ........ Merged revisions 360309 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360310 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13Add ability for chan_dahdi ISDN to block connected line updates per span.Richard Mudgett
Added new chan_dahdi.conf colp_send option parameter to block connected line updates per span. (closes issue ASTERISK-17025) Reported by: Michael Smith git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-02Remove ISDN hold restriction for non-bridged calls.Richard Mudgett
The check if an ISDN call is bridged before it could be placed on hold is not necessary and is overly restrictive. The check was originally done to prevent problems with call transfers in case a user tried to transfer a call connected to an application to another call connected to an application. The ISDN transfer code has not required this restriction for quite some time because ECT could transfer any two active calls to each other. * Remove ISDN hold restriction for calls connected to applications. * Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and AST_CONTROL_UNHOLD instead of generating a warning message. (closes issue ASTERISK-19388) Reported by: Birger Harzenetter Tested by: rmudgett ........ Merged revisions 357894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357895 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-01Opaquify ast_channel typedefs, fd arrays, and softhangup flagTerry Wilson
Review: https://reviewboard.asterisk.org/r/1784/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-29Opaquify ast_channel structs and listsTerry Wilson
Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28Use more reasonable cause code when rejecting incoming call waiting calls.Richard Mudgett
(closes issue ASTERISK-19397) Reported by: Birger Harzenetter Patches: nochannel-cause.patch (license #5870) patch uploaded by Birger Harzenetter ........ Merged revisions 357407 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357408 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20ast_channel opaquification of pointers and integral typesTerry Wilson
Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-18push 'outgoing' flag from sig_XXX up to chan_dahdiAlec L Davis
'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ ........ Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355851 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-13Opaquify char * and char[] in ast_channelTerry Wilson
Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-02Restore the 'w' modifier support for ISDN spans. Dial(DAHDI/g0/1234w888)Richard Mudgett
This feature also causes the sending complete ie to be sent for switch types that do not automatically send the ie. (EuroISDN/ETSI) The main difference between dialing Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the sending of the sending complete ie. (closes issue ASTERISK-19176) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 353867 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353868 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-01Constify some more channel driver technology callback parameters.Richard Mudgett
Review: https://reviewboard.asterisk.org/r/1707/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24Opaquify channel stringfieldsTerry Wilson
Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09Replace direct access to channel name with accessor functionsTerry Wilson
There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-17Remove dead code since pri_grab() can never fail.Richard Mudgett
Dead code makes programmers sick. I am sick of looking at it. ........ Merged revisions 345546 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345558 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-15Fix typo in sig_pri using wrong structure name.Richard Mudgett
It is fortunate that the typo does not alter generated code since the e->restart.channel and e->ring.channel members are in the same position. (closes issue ASTERISK-18868) Reported by: zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by zvision ........ Merged revisions 345370 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345371 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-25Change D-channel warning to be less confusing on non-NFAS setups.Richard Mudgett
The "No D-channels available! Using Primary channel as D-channel anyway!" WARNING message has been confusing on non-NFAS setups. The message refers to things that are NFAS specific. * Changed the warning to several different warnings to be more accurate for the situation and less confusing as a result: "No D-channels up! Switching selected D-channel from X to Y.", "No D-channels up!", and "D-channel is down!". ........ Merged revisions 342484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342485 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342486 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338323 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338323 | rmudgett | 2011-09-28 17:36:57 -0500 (Wed, 28 Sep 2011) | 12 lines Merged revisions 338322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) | 5 lines Make duplicate call ptr warning message more helpful. * Adds the value of the call ptr to the duplicate call ptr message to help trace why there is a duplicate call ptr. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337721 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337721 | rmudgett | 2011-09-22 16:37:41 -0500 (Thu, 22 Sep 2011) | 25 lines Merged revisions 337720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) | 18 lines Made ISDN not add numbering plan prefix strings to empty numbers. When the Caller-ID is restricted, the expected behavior is for the Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto the Caller-ID number even if it is restricted (empty) causing the Caller-ID to be the national prefix rather than blank. This behavior was lost when sig_pri was extracted from chan_dahdi. * Made not add prefix strings to empty connected line, calling, and ANI number strings. (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches: jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Kris Shaw ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-19Merged revisions 336570 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336570 | rmudgett | 2011-09-19 10:32:00 -0500 (Mon, 19 Sep 2011) | 11 lines Merged revisions 336569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA AST-675 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-09Merged revisions 335078 via svnmerge from Matthew Jordan
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-17Merged revisions 332265 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332265 | rmudgett | 2011-08-17 11:01:29 -0500 (Wed, 17 Aug 2011) | 33 lines Merged revisions 332264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) | 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-28Merged revisions 330051 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330051 | rmudgett | 2011-07-28 12:10:37 -0500 (Thu, 28 Jul 2011) | 29 lines Merged revisions 330050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r330050 | rmudgett | 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) | 22 lines Merged revisions 330033 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and outgoing call legs of a data call are using different formats: a-law, u-law. When the call is bridged, the media stream is run through translation to convert the media formats. The translation is bad for data calls. * Make incoming call that does not explicitly specify u-law or a-law use the DAHDI channel's default law. The outgoing call always uses the default law from the DAHDI channel. (closes issue ABE-2800) Patches: jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett .......... ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21Merged revisions 329257 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines s/1.10/10.0/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-17Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message.Richard Mudgett
The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how much of the current time is put in Q931_IE_TIME_DATE. * Send date/time ie never. * Send date/time ie date only. * Send date/time ie date and hour. * Send date/time ie date, hour, and minute. * Send date/time ie date, hour, minute, and second. * Send date/time ie default: Libpri will send date and hhmm only when in NT PTMP mode to support ISDN phones. (closes issue #19221) Reported by: kenner JIRA SWP-3396 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-13Merged revisions 318783 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) | 14 lines PRI early media won't ring. And another way to pass early media. Don't indicate that there is inband information present, just assume that the B channel is connected. * Restore clearing the dialing flag Rx squelch unconditionally when a PROCEEDING message comes in. (closes issue #19268) Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by rmudgett (license 664) Tested by: tbsky ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-10Merged revisions 318499 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) | 15 lines Unable to pickup DAHDI/PRI call because call state is reported as DIALING. The channel state is not updated to RINGING when an ALERTING message is received. Regression caused when sig_pri.c (also sig_ss7.c) extracted from chan_dahdi.c. * Added missing channel state update to RINGING when the AST_CONTROL_RINGING frame is queued for ISDN and SS7. (closes issue #19257) Reported by: alecdavis Patches: issue19257_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-09Merged revisions 318231 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318231 | rmudgett | 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) | 41 lines Don't get early media for ISDN on outgoing calls. It looks to be a long-standing misinterpretation of the progress indicator ie values: 1 - Call is not end-to-end ISDN; further call progress information may be available in-band. 8 - In-band information or an appropriate pattern is now available. Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled as early media probably because the meaning of the second half of it's description was overlooked. * Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path. (closes issue #18868) Reported by: isrl Patches: issue18868_19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: satish_lx .......... No inband progress on PRI_EVENT_RINGING even if inband flag set. My ISDN-PRI provider sends an ALERTING with "Inband information or appropriate pattern now available", but Asterisk only generates and passes the RING to the SIP extension, not the inband message. Unfortunately, the inband message is not a ringback tone but a prompt that says the number is not in service. The SIP extension then hears two rings and the call is hungup which confuses the caller. * Post an AST_CONTROL_PROGRESS as well as opening the media path if inband audio is indicated with an ALERTING message. (closes issue #19246) Reported by: cristiandimache Patches: issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: cristiandimache ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-03Merged revisions 316224 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316224 | rmudgett | 2011-05-03 14:18:30 -0500 (Tue, 03 May 2011) | 16 lines The dahdi_hangup() call does not clean up the channel fully. After dahdi_hangup() has supposedly hungup an ISDN channel there is still traffic on the S0-bus because the channel was not cleaned up fully. Shuffled the hangup code to include some missing cleanup. Also fixed some code formatting in the area. I think the primary missing clean up code was the call to tone_zone_play_tone() to turn off any active tones on the channel. (closes issue #19188) Reported by: jg1234 Patches: issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested by: jg1234 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-21Implement AMI action PRIShowSpans.Richard Mudgett
PRIShowSpans works like the AMI action DAHDIShowChannels but for PRI spans. It is similar to the CLI command "pri show spans". (closes issue #15980) Reported by: dwery git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-21Simplify sig_pri.c:build_status().Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-18Problems with ISDN MWI to phones.Richard Mudgett
The "controlling user number" is always the number of the voice mail box which is identical with the subscriber number itself. This number which is listed in the ISDN phone MWI menu cannot be called back to contact the voice mail box. The controlling user number should be made configurable. JIRA ABE-2738 JIRA SWP-2846 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-08Add private lock deadlock avoidance callback to PRI and SS7.Richard Mudgett
Factor out the equivalent function for analog. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-07Merged revisions 313001 via svnmerge from Alec L Davis
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313001 | alecdavis | 2011-04-07 22:19:31 +1200 (Thu, 07 Apr 2011) | 13 lines Fix ISDN calling subaddr User Specified Odd/Even Flag Calculation of the Odd/Even flag was wrong. Implement correct algo, and set odd/even=0 if data would be truncated. Only allow automatic calculation of the O/E flag, don't let dialplan influence. (closes issue #19062) Reported by: festr Patches: bug19062.diff2.txt uploaded by alecdavis (license 585) Tested by: festr, alecdavis, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-05Merged revisions 312949 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312949 | rmudgett | 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011) | 6 lines Crash if ISDN span layer 1 is down on initial load. Regression from -r312575 B channel shifting during negotiation. * Also combine updating the alarm flag with clearing the resetting flag. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-04Remove the channel parameter from sig_pri_handle_subcmds().Richard Mudgett
It was only used in a debug message and may not be correct anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-04Merged revisions 312575 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312575 | rmudgett | 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) | 52 lines Merged revisions 312574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines Merged revisions 312573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines Issues with ISDN calls changing B channels during call negotiations. The handling of the PROCEEDING message was not using the correct call structure if the B channel was changed. (The same for PROGRESS.) The call was also not hungup if the new B channel is not provisioned or is busy. * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are using the correct structure and B channel. If there is any problem with the operations then the call is now hungup with an appropriate cause code. * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the correct structure by looking for the call and not using the channel ID. NOTIFY is an exception with versions of libpri before v1.4.11 because a call pointer is not available for Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct structure by looking for the call and not using the channel ID. (closes issue #18313) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-18Merged revisions 311297 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) | 12 lines Race condition when ISDN CallRerouting/CallDeflection invoked. The queued AST_CONTROL_BUSY could sometimes be processed before the call_forward dial string is recognized. * Moved setting the call_forwarding dial string after sending a response to the initiator and just queue an empty frame to wake up the media thread instead of an AST_CONTROL_BUSY. * Added check for empty rerouting/deflection number and respond with an error. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-08Merged revisions 309994 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) | 1 line Make pri parameter description consistent. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-04Merged revisions 309445 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-23Merged revisions 308622 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails. (closes issue #18874) Reported by: cmaj Patches: patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830) JIRA SWP-3172 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-15Add CLI "pri show channels" command.Richard Mudgett
List the current mapping of DAHDI B channels to Asterisk channel names and which calls are on hold or call-waiting. Calls on hold or call-waiting are not associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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