259 Commits (34cce24a9c10b177d93650b6003c82d80cd9aa4d)

Author	SHA1	Message	Date
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cf7bbcc4c6	Expand the caller ANI field to an ast_party_id ... Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring ... The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	c613897d1c	Fix various typos reported by Lintian ... (Also fix the typos in the comments) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b5d5cc565f	Enhancements to connected line and redirecting work. ... From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	e2336b73ef	Merged revisions 262657,262660 from ... https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed, 12 May 2010) | 4 lines Forgot some conditionals around the callrerouting facility help text. JIRA ABE-2223 .......... r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010) | 22 lines Add mISDN Call rerouting facility for point-to-point ISDN lines (exchange line) In the case of ISDN point-to-multipoint (multidevice) you can use the mISDN "facility calldeflect" application for call diversions from external (PSTN) to external (PSTN). In that case this is the only way to get rid of the two call legs to the PBX and let the calling number at the C party become the number of the A party. In the case of ISDN point-to-point (exchange line) the call deflection facility may not be used. Instead a call rerouting facility has to be used. This patch for chan_misdn.c is an extension to realize this service (facility rerouting application). It can accept either spelling: "callrerouting" or "callrerouteing". The patch is tested towards Deutsche Telekom and requires a modified version of mISDN from Digium, Inc. Patches: misdn_rerouteing_corrected.patch (Slightly modified.) JIRA ABE-2223 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	a5a0a5f867	Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number. ... SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	57ee669d9f	Merged revisions 247910 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ef9be94b35	Change all refererences to 1.6.3 to be 1.8, since that will be the next feature release ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	45ad82fda3	Merged revisions 228078 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out an ast_frame. (closes issue #16041) Reported by: francesco_r ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. ... Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9456ab2724	Deadlock in channel masquerade handling ... Channels are stored in an ao2_container. When accessing an item within an ao2_container the proper locking order is to first lock the container, and then the items within it. In ast_do_masquerade both the clone and original channel must be locked for the entire duration of the function. The problem with this is that it attemptes to unlink and link these channels back into the ao2_container when one of the channel's name changes. This is invalid locking order as the process of unlinking and linking will lock the ao2_container while the channels are locked!!! Now, both the channels in do_masquerade are unlinked from the ao2_container and then locked for the entire function. At the end of the function both channels are unlocked and linked back into the container with their new names as hash values. This new method of requiring all channels and tech pvts to be unlocked before ast_do_masquerade() or ast_change_name() required several changes throughout the code base. (closes issue #15911) Reported by: russell Patches: masq_deadlock_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, atis (closes issue #15618) Reported by: lmsteffan Patches: deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671) Tested by: lmsteffan, dvossel Review: https://reviewboard.asterisk.org/r/387/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	49b90d5e61	Merged revisions 222691 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	a305b52fc8	Send a generic return result when we receive a CallDeflection facility message in chan_misdn. ... ETSI 300-196 implies that a facility return result without arguments does not have the operation-value. This fact implies for ETSI that you can only use the invoke-id to match requests with responses. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	642bec4d6f	AST-2009-005 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ed2a3cedd1	Merged revisions 209759 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines Minor changes inspired by testing with latest GCC. The latest GCC (what will become 4.5.x) has a few new warnings, that in these cases found some either downright buggy code, or at least seriously poorly designed code that could be improved. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	bcff592839	Merged 207316 from ... https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r207316 | rmudgett | 2009-07-17 23:05:05 -0500 (Fri, 17 Jul 2009) | 20 lines Fixed incoming calls being matched to MSNs without type-of-number prefix added. For an incoming ISDN call the dialed.number is incorrectly matched against the configured MSNs in misdn.conf. The numbers passed to the dialplan include the configured prefix for the dialed.number_type, whereas the check against the configured MSNs (to decide if the call is accepted at all), is executed without the configured prefix. e.g., dialed.number = 241168020, TON = national, configured national prefix is "0". (This is the TON which is used by ISDN providers in the Netherlands.) In chan_misdn.c:cb_events() in case EVENT_SETUP the call to misdn_cfg_is_msn_valid() uses the unnormalized number 241168020, but 57 lines later the call to read_config() adds the prefix, and the dialed.number is now 0241168020, which is then used in the dialplan. misdn_cfg_is_msn_valid() must use the normalized number, too. JIRA ABE-1912 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	58b440bc29	Merged revisions 206487 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines Fixes several call transfer issues with chan_misdn. * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	c90a8c0921	Merged revisions 206284 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	a894c33cb3	Merged revisions 204834 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. ... 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: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	2ce62b35cf	Make outgoing_colp=2 misdn.conf port parameter not send redirecting or transfer messages. ... If the outgoing_colp parameter is set to not send COLP information, then it does not make sense to send redirecting or transfer messages announcing new COLP information that is blocked. The service provider may supply the listed number for that line when it passes the messages to the next hop. Why tell the switch that these events happened when the information is otherwise suppressed? Also blocked the number of previous redirects that may have occurred to calls going out the port when outgoing_colp is 2. Follow on to JIRA ABE-1853. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4379249674	Convert a number of global module variables to 'static'. ... These modules all contained variables that are module-global but not system-global, but were not marked 'static'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	63e4b99e79	Make chan_misdn compile. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	678045fb43	Merged revisions 196116 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 lines Fix a bug where using immediate with mISDN caused a cause code of 16 to get sent back instead of 1 if the 's' extension did not exist. (closes issue #12286) Reported by: lmamane ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) ... This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	7872538b83	Add outgoing_colp misdn.conf port parameter. ... Select what to do with outgoing COLP information on this port. 0 - Send out COLP information unaltered. (default) 1 - Force COLP to restricted on all outgoing COLP information. 2 - Do not send COLP information. outgoing_colp=0 Also fixed sending the EctInform message so it always has the required redirectionNumber parameter when the status is active. JIRA ABE-1853 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	1c988d8996	add 'const' qualifiers in various places where they should have been ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	3b01ff719c	Merged revisions 193613 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines Sent wrong message to clear a call we started if the other end has not responed yet. In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet), it is not allowed to clear the call with RELEASE_COMPLETE. It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193614 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	90f76fcfba	Merged revisions 193050 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines Give a more helpful message when an incoming call's dialed extension does not match. Added the dialed extension and context to the chan_misdn messages warning that the dialed number cannot be matched in the dialplan. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	d35fd35ae3	Outgoing PTP redirected calls did not wait for the COLR from the redirected-to party. ... For outgoing PTP redirected calls, you now need to use the inhibit(i) option on all of the REDIRECTING statements before dialing the redirected-to party. You still have to set the REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The PTP call will update the redirecting-to presentation when it becomes available and queue the redirecting update to the calling channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	89d06c7759	Make PTP DivertingLegInformation3 message behavior closer to the specifications. ... * Wait for a DivertingLegInformation3 message after receiving a DivertingLegInformation1 message to complete the redirecting-to information before queuing a redirecting update to the other channel. * A DivertingLegInformation2 message should be responded to with a DivertingLegInformation3 when the COLR is determined. If the call could or does experience another redirection, you should manually determine the COLR to send to the switch by setting REDIRECTING(to-pres) to the COLR and setting REDIRECTING(to-num) = ${EXTEN}. * A DivertingLegInformation2 message must have an original called number if the redirection count is greater than one. Since Asterisk does not keep track of this information, we can only indicate that the number is not available due to interworking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	c95c065903	There is no need to use the struct ast_party_connected_line.source update values. ... The messages sent by a technology when a connected line update is received are best determined by the current call state of the channel. The struct ast_party_connected_line.source value is really only useful as a possible tracing aid. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c878b21887	Fix a build error. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cba19c8a67	Convert the ast_channel data structure over to the astobj2 framework. ... There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	6bb2b6c096	Added CCBS/CCNR Party A support and enhanced COLP support. ... This change adds the following features to chan_misdn: * CCBS/CCNR Party A support for PTMP and PTP modes. * Enhances COLP support for call diversion and explicit call transfer. These enhanced features require a modified version of mISDN. The latest modified mISDN v1.1.x based version is available at: http://svn.digium.com/svn/thirdparty/mISDN/trunk http://svn.digium.com/svn/thirdparty/mISDNuser/trunk Taged versions of the modified mISDN code are available under: http://svn.digium.com/svn/thirdparty/mISDN/tags http://svn.digium.com/svn/thirdparty/mISDNuser/tags Review: http://reviewboard.digium.com/r/218/ Merged from team/rmudgett/misdn_facility branch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	13af9ed74c	Merged revisions 188833,189134 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled. Leave jitter setting alone. JIRA ABE-1835 ........ r189134 | rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines Modifed/added some debug messages. JIRA ABE-1835 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	b0cfe6e1f2	Miscellaneous minor changes to chan_misdn. ... * Miscellaneous spacing and comment changes. * Minor code rearangements. * Miscellaneous doxygen comments. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	b89dce07b4	Make chan_misdn_log() avoid generating the log message if logging is disabled. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	6f53ed4c67	This commit introduces COLP/CONP and Redirecting party information into Asterisk. ... The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	9a6bf5f9c6	Removed trailing whitespace in chan_misdn files. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4ec301360c	Merge a large set of updates to the Asterisk indications API. ... This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	f029e410ec	A further correction: cast the sizeof to an int. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	5918a38b98	my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2 didn't like the \%ld and issued a warning, breaking my dev-mode build. This fixes it. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	de535452cd	build in dev mode ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9c10ccd9fd	Mostly spacing changes; no functionality change at all. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	55b4eedd5d	Janitor patch for chan_misdn (make channel variable access safe) ... (closes issue #12887) Reported by: pputman Patches: chan_misdn_threadsafe.patch uploaded by pputman (license 81) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ef6ad2b53c	Merged revisions 168561 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	17d56d409b	Spacing change ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	4d1500edbd	Merged revisions 168191 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009) | 3 lines * Fix for JIRA AST-175/ABE-1757 * Miscellaneous doxygen comments added. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago