361 Commits (04224c20362f94f99886d6b51adb7b90db62855f)

Author	SHA1	Message	Date
Richard Mudgett	d37ea04b2f	Merged revisions 335851 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) | 11 lines Fixed cut-n-paste regression using the wrong variable. Fixes the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration. (closes issue ASTERISK-18496) Reported by: Sean Darcy Patches: jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Sean Darcy, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	4e57652651	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/branches/10@335078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2723aba2fa	Merged revisions 334012 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) | 23 lines No DAHDI channel available for conference, user introduction disabled. The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards Patches: jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c33ad6fc76	Merged revisions 332503 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) | 8 lines CRC4 in "dahdi show status" gives wrong impression to T1 users Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in more situations without confusing users, especially since T1 lines use CRC6 instead of CRC4. (closes issue AST-471) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cbfbbbeb32	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/branches/10@332265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	41549fcd66	Merged revisions 331955 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) | 13 lines Fix some minor chan_dahdi config load issues. * Address chan_dahdi.conf dahdichan option todo item about needing line number. * Make ignore_failed_channels option also apply to dahdichan option. * Don't attempt to create a default pseudo channel if the chan_dahdi.conf channel/channels option is not allowed. * Add a similar check for dahdichan in normal chan_dahdi.conf sections as is done in users.conf. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4e9515bab0	Merged revisions 331771 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) | 8 lines Suppress warning message when using DAHDITransfer or DAHDIHangup. * The fake event should only be processed by the channel that currently owns the private and not the associated call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a9a9bb4bda	Merged revisions 331714 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) | 22 lines AMI actions DAHDIHangup and DAHDITransfer have no effect. The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI channel. These two AMI actions are highly specialized to analog channels and appear to make the channel behave like a jack port for headsets. * Made the faked DAHDI event get processed before a normal media stream read in dahdi_read() instead of trying to trigger an exception read by setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long ago that changed how AST_FLAG_EXCEPTION is processed in the core. Unfortunately, the faked DAHDI events no longer worked when that happened. * Updated the DAHDI AMI action documentation for the following actions: DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and DAHDIRestart. * Made use sscanf() instead of atoi() for better error checking of the DAHDIChannel header string. JIRA AST-620 JIRA SWP-3616 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	82a2f7541e	Merged revisions 330705 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) | 10 lines Call pickup broken for DAHDI channels when beginning with # The call pickup feature did not work on DAHDI devices for anything other than feature codes beginning with * since all feature codes in chan_dahdi were originally hard-coded to begin with *. This patch is also applied to chan_dahdi.c to fix this bug with radio modes. (closes issue AST-621) Review: https://reviewboard.asterisk.org/r/1336/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	94bbb01fdd	s/1.10/10.0/ ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	46dd023878	Merged revisions 329203 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines Document parkinglot in chan_dahdi.conf.sample. * Document existing feature in chan_dahdi.conf.sample. * Remove some dead code related to the parkinglot option. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7caa2349af	Merged revisions 328209 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	1353e83531	Merged revisions 327044 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327044 | russell | 2011-07-08 10:28:44 -0500 (Fri, 08 Jul 2011) | 2 lines Resolve some set-but-unused-variable warnings. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	7d179abfd4	Merged revisions 326411 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines Add the attribute "type" to each "<use>" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	70be58c1a7	Merged revisions 325212 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) | 7 lines Use the device name and not the channel name to initialize the device state. Correct ASTERISK-11323 implementation as I don't see how it ever worked as claimed when it used the channel name and not the device name. (issue ASTERISK-11323) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	6f74606fda	Merged revisions 324174 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17 Jun 2011) | 5 lines Add header string to libpri debug output. Add header string to libpri debug output so the libpri output can be found/extracted easier from huge debug trace files. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5257a915a8	Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message. ... 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	14 years ago
Russell Bryant	0938974902	Merged revisions 317478 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines Fix some consistency issues with jitterbuffer config. Store the defaults noted in the sample config files in the jitterbuffer config data structure. This makes the CLI commands that output these settings show the right thing. Also only show the settings that are relevant in the settings CLI commands, based on which jitterbuffer is selected and whether it's enabled. (closes issue #19083) Reported by: rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	810b9c8879	Merged revisions 316224 via svnmerge from ... 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	14 years ago
Alec L Davis	73d8795841	Merged revisions 315001 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315001 | alecdavis | 2011-04-23 10:59:18 +1200 (Sat, 23 Apr 2011) | 12 lines chan_dahdi: Can't return to normal ring after distinctive ring on FXS clear a previous distinctivering pattern before each new call (closes issue #18985) Reported by: bromont Patches: bug18985.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, bromont ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0f1ff9141e	Implement AMI action PRIShowSpans. ... 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	14 years ago
Richard Mudgett	c2676dc9dc	Merged revisions 314732 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314732 | rmudgett | 2011-04-21 17:38:44 -0500 (Thu, 21 Apr 2011) | 1 line Correct DAHDIShowChannels XML documentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	37274c73ee	Problems with ISDN MWI to phones. ... 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	14 years ago
Richard Mudgett	4f8d56a824	Merged revisions 313780 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14 Apr 2011) | 20 lines Leftover debug messages unconditionally sent to the console. Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config option enabled outputs the following debug messages unconditionally: Dialing T1847555121 on 1 Dialing www2w on 1 * Made debug messages in my_dial_digits() normal debug messages that do not get output unless enabled. * Reworded some debug messages in my_dial_digits() to be clearer. * Replace strncpy() with ast_copy_string() in my_dial_digits() which does the same job better. (closes issue #18847) Reported by: vmikhelson Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	b8b1d085db	Add 'description' field for CLI and Manager output ... (closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	a6695b84ce	Merged revisions 313435 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 also went ahead and fixed the problem it introduces before committing. ........ r313435 | jrose | 2011-04-12 13:44:44 -0500 (Tue, 12 Apr 2011) | 1 line fixing stupid mistake with putting code before variable declaration ........ Merged revisions 313433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | 14 lines reload Chan_dahdi memory leak caused by variables chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would stay in the dahdi_pvt structs for individual channels (causing them to just continue adding the new ones to the list) and also there was a memory leak causes by the conf objects. This patch resolves both of these by using ast_variables_destroy during the loading process. (closes issue #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by jrose (license 1225) Tested by: tilghman, jrose Review: https://reviewboard.asterisk.org/r/1170/ ........ ........ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	bc907695bf	Merged revisions 313190 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313190 | rmudgett | 2011-04-11 10:40:30 -0500 (Mon, 11 Apr 2011) | 39 lines Merged revisions 313189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313189 | rmudgett | 2011-04-11 10:32:53 -0500 (Mon, 11 Apr 2011) | 32 lines Merged revisions 313188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) | 25 lines Stuck channel using FEATD_MF if caller hangs up at the right time. The cause was actually a caller hanging up just at the end of the Feature Group D DTMF tones that setup the call. The reason for this is a "guard timer" that's implemented using ast_safe_sleep(100). If the caller happens to hang up AFTER the final tone of the DTMF string but BEFORE the end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero. This causes the code to bounce to the end of ss_thread(), but it does NOT tear down the call properly. This should be a rare occurrence because the caller has to hang up at EXACTLY the right time. Nonetheless, it was happening quite regularly on the reporter's system. It's not easily reproducible, unless you purposely increase the guard-time to 2000 or more. Once you do that, you can reproduce it every time by watching the DTMF debug and hanging up just as it ends. Simply add an ast_hangup() before goto quit. (closes issue #15671) Reported by: jcromes Patches: issue15671.patch uploaded by pabelanger (license 224) Tested by: jcromes ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ce17f956dc	Add private lock deadlock avoidance callback to PRI and SS7. ... Factor out the equivalent function for analog. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	698a356737	Merged revisions 312949 via svnmerge from ... 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	14 years ago
Richard Mudgett	e1ceb52b51	Merged revisions 312575 via svnmerge from ... 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	14 years ago
Jonathan Rose	759bf6b840	Fixing bad line break from 312384 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	846cfa0ef0	New Feature for chan_dahdi. 4 length pattern matching. ... In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s ntax remains the same and the method used to track the pattern history will only change when using the length 4 patterns. (closes issue SWP-3250) Code: jrose rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8dce4dbe2a	Merged revisions 311874 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311874 | rmudgett | 2011-03-29 20:56:05 -0500 (Tue, 29 Mar 2011) | 1 line Update some setup_dahdi_int() comments. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311875 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	6de1332214	Merged revisions 309808 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines Merged revisions 309251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Moises Silva	0f207dce6e	Merged revisions 309720 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar 2011) | 6 lines Fix caller id passed to openr2_chan_make_call (closes issue #18894) Reported by: malufrj Tested by: moy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	928ec2b990	Merged revisions 309445 via svnmerge from ... 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	14 years ago
Richard Mudgett	b79adb645e	Add more verbage to CLI command 'pri show channels' usage. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4a48600231	Add CLI "pri show channels" command. ... 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	14 years ago
Richard Mudgett	b2ef13cb60	Merged revisions 307879 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines No response sent for SIP CC subscribe/resubscribe request. Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	49feb747ba	Pass a MCID request to the bridged channel. ... Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	484f9bec0a	Ignore voice frames in chan_dahdi native bridging. Hardware is handling them. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a8aeb04a9f	Add ISDN display ie text handling options to chan_dahdi.conf. ... The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	3556e4c2d4	Replace ast_log(LOG_DEBUG, ...) with ast_debug() ... (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields ... This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ecdbb3d1d9	Merged from revision 304341 ... 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: https://origsvn.digium.com/svn/asterisk/trunk@304385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	15605be78b	Merged revisions 304150 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304150 | rmudgett | 2011-01-26 13:39:35 -0600 (Wed, 26 Jan 2011) | 16 lines Merged revisions 304149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600 (Wed, 26 Jan 2011) | 9 lines Merged revisions 304148 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, 26 Jan 2011) | 2 lines Update documentation for DAHDISendCallreroutingFacility() application. .......... ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7889af7cab	Merged revisions 303771 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	54f6c31a27	Merged revisions 303467 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303467 | qwell | 2011-01-24 11:20:03 -0600 (Mon, 24 Jan 2011) | 22 lines Merged revisions 303285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines Reset configuration before parsing users.conf. Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	95f5dc6644	Temporarily revert r303288 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	4272837ead	Merged revisions 303286 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303286 | qwell | 2011-01-21 15:50:11 -0600 (Fri, 21 Jan 2011) | 22 lines Merged revisions 303285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines Reset configuration before parsing users.conf. Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago