asterisk

Commit Graph

Author	SHA1	Message	Date
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
David Vossel	cddc244c97	fix iax_frame double free Very unfortunate things happen if we add an iax_frame to the frame queue and let go of the lock before scheduling the frame's transmit... There is a race condition that exists where the frame can be removed from the frame_queue and freed before the transmit is scheduled if we do not hold on to that lock. This results in a freed frame being scheduled for transmit later. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	274eb8960c	Fix inverted logic in cli command: ss7 set debug on/off git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c26cd3aaac	Fix an off by one error that causes a crash. Thanks to Raymond Burke for pointing it out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	093dbfdd52	Don't crash when destroying chan_dahdi pseudo channels. Must do a deep copy of the cc_params in duplicate_pseudo(). Otherwise, when the duplicate pseudo channel is destroyed, it frees the original pseudo channel cc_params. The original pseudo channel is then left with a dangling pointer for when the next duplicated pseudo channel is created. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262743 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
Paul Belanger	4b1d9f85a7	Convert to AST_CLI_YESNO and AST_CLI_ONOFF Clean up chan_sip.c to use new AST_CLI functions (closes issue #17287) Reported by: pabelanger Patches: issue17287.patch uploaded by pabelanger (license 224) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	9534f72cb0	Dialing an invalid extension causes incomplete hangup sequence. Revision -r1489 of the libpri 1.4 branch corrected a deviation from Q.931 Section 5.3.2. However, this resulted in an unexpected behaviour change to the upper layer (Asterisk). This change uses pri_hangup_fix_enable() to follow Q.931 Section 5.3.2 call hangup better if the version of libpri supports it. (issue #17104) Reported by: shawkris Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	9c012b460f	Improve logging information for misconfigured contexts (closes issue #17238) Reported by: pprindeville Patches: chan_sip-bug17238.patch uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	351e0e90c5	fixes crash in chan_console There is a race condition between console_hangup() and start_stream(). It is possible for console_hangup() to be called and then the stream thread to begin after the hangup. To avoid this a check in start_stream() to make sure the pvt-owner still exists while the pvt lock is held is made. If the owner is gone that means the channel hung up and start_stream should be aborted. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	356b090875	Fix deadlock in sig_pri when hanging up. The pri_dchannel thread currently violates locking order by locking the private and then attempting to queue a frame, which needs to lock the channel. Queueing a frame is unneccesary though and is actually a regression since sig_pri. All the places that currently use ast_softhangup_nolock now will just set the softhangup value directly as before. (closes issue #17216) Reported by: lmsteffan Patches: bug17216.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	6f03bf4a42	Some code optimizations. * Made more places use pri_queue_control() instead of pri_queue_frame() and a local frame variable. * Made pri_queue_frame() use sig_pri_lock_owner(). pri_queue_frame() no longer releases the libpri access lock unless it is required. * Made the pri_queue_frame() and pri_queue_control() parameter list similar to sig_pri_lock_owner(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ba9b0d95e6	Permit more lines within a SIP body to be parsed. The example given within the related issue showed 120 lines, which was mostly a result of the body being XML. (closes issue #17179) Reported by: khw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	6b2e51b1ed	When failing to configure, don't destroy 'cfg' twice Fixes a crash when some config section had an incorrect channel config. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	f0a7adb309	Avoid a crash on SS7 channels. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f16625b7a0	fixes sip native transfer The Refer-To header field containing the Replaces header in the URI was not being decoded properly. This caused invalid parsing between the caller id field and the domain resulting in a failed transfer. (closes issue #17284) Reported by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	35eeb71ead	Merged revisions 261274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261274 \| pabelanger \| 2010-05-05 12:42:22 -0400 (Wed, 05 May 2010) \| 12 lines Registration fix for SIP realtime. Make sure realtime fields are not empty. (closes issue #17266) Reported by: Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis, sberney Review: https://reviewboard.asterisk.org/r/643/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a6ea125e7c	Prevent unnecessary warnings when getting rtpsource or rtpdest. If a recognized media type was present, but the media type was not enabled for the channel, then a warning would be emitted. For instance, attempting to get CHANNEL(rtpsource,video) on a call with no video would cause a warning message to appear. With this change, the warning will only appear if the stream argument is not recognized as being a media type that can be specified. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	159f0d4b24	The inalarm flag is not passed up from the sig_analog and sig_pri submodules. The CLI "dahdi show channel" command was not correctly reporting the InAlarm status. The inalarm flag is now consistently passed between chan_dahdi and submodules. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	6dd80de93d	Merged revisions 260434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260434 \| jpeeler \| 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) \| 11 lines Ensure channel state is not incorrectly set in the case of a very early answer. The needringing bit was being read in dahdi_read after answering thereby setting the state to ringing from up. This clears needringing upon answering so that is no longer possible. (closes issue #17067) Reported by: tzafrir Patches: needringing.diff uploaded by tzafrir (license 46) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	00a1a96ff5	Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7, and MFCR2 users. Created SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS SIG_MFCR2_MAX_CHANNELS Also fixed the declaration of pollers[] in mfcr2_monitor(). It was dimensioned to the number of bytes in struct dahdi_mfcr2.pvts[] and not to the same dimension of the struct dahdi_mfcr2.pvts[]. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	a3ce3441bb	Merged revisions 260195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260195 \| rmudgett \| 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) \| 26 lines DTMF CallerID detection problems. The code handling DTMF CallerID drops digits on long CallerID numbers and may timeout waiting for the first ring with shorter numbers. The DTMF emulation mode was not turned off when processing DTMF CallerID. When the emulation code gets behind in processing the DTMF digits it can skip a digit. For shorter numbers, the timeout may have been too short. I increased it from 2 seconds to 4 seconds. Four seconds is a typical time between rings for many countries. (closes issue #16460) Reported by: sum Patches: issue16460.patch uploaded by rmudgett (license 664) issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA AST-334 JIRA SWP-901 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9e1b6c7236	Don't override peer context with domain context. (closes issue #17040) Reported by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347) Tested by: pprindeville Review: https://reviewboard.asterisk.org/r/565/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6722251986	Merged revisions 259858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259858 \| dvossel \| 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) \| 33 lines resolves deadlocks in chan_local Issue_1. In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock avoidance is done when the channel to hangup is the outbound chan_local channel, but when it is not the outbound channel we have an issue... We attempt to do deadlock avoidance only on the tech pvt, when both the tech pvt and the pvt->owner are locked coming into that loop. By never giving up the pvt->owner channel deadlock avoidance is not entirely possible. This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is used in ast_activate_generator() to queue a frame on the channel and make the channel's read function get called. This function is used in ast_activate_generator() while the channel is locked, which mean's the channel will have a lock both from the generator code and the frame_queue code by the time it gets to chan_local.c's local_queue_frame code... local_queue_frame contains some of the same crazy deadlock avoidance that local_hangup requires, and this recursive lock prevents that deadlock avoidance from happening correctly. This patch removes ast_prod() from the channel lock so only one lock is held during the local_queue_frame function. (closes issue #17185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	0da0e3856c	Be more explicit about field naming in a test. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	4e3269c60d	Merged revisions 259531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259531 \| rmudgett \| 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) \| 11 lines DAHDI "WARNING" message is confusing and vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success" Changed the warning to "Failed to decode CallerID on channel 'name'". The message before it is likely more specific about why the CallerID decode failed. SWP-501 AST-283 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	3e04d6fe8e	Merged revisions 259270 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259270 \| rmudgett \| 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) \| 14 lines hidecalleridname parameter in chan_dahdi.conf Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	3bda444f1e	Re-fix dahdi_request() iflist locking since CCSS merged. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	1b62cf14e4	Small error in the T.140 RTP port verbose log. (closes issue #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	bb2fa21ac1	IAXpeers output now matches SIPpeers format for manager (AMI). (closes issue #17100) Reported by: secesh Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/594/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f2b8561a5a	fixes issue with double "sip:" in header field This is a clear mistake in logic. Future discussions about how to avoid having to handle uri's like this should take place in the future, but this fix needs to go in for now. (closes issue #15847) Reported by: ebroad Patches: doublesip.patch uploaded by ebroad (license 878) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ba81ee6d28	Make sure to fail a monitor if we receive a negative response for a CC SUBSCRIBE. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Dwayne M. Hubbard	afedb856ae	Enable PRI SERVICE message support in chan_dahdi for the 'national' switchtype Revision 1072 of libpri added SERVICE message support for the 'national' switchtype. The attached patch enables the use of 'pri service' CLI commands on dahdi channels that are configured for the 'national' switchtype. (closes issue #17142) Reported by: dhubbard Patches: dw-ni2.patch uploaded by dhubbard (license 733) Tested by: elguero, dhubbard Review: https://reviewboard.asterisk.org/r/612/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	84d0b95def	Merged revisions 257467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257467 \| tilghman \| 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) \| 13 lines Don't recreate peer, when responding to a repeated deregistration attempt. When a reply to a deregistration is lost in transmit, the client retries the deregistration. Previously, this would cause a realtime/autocreate peer to be loaded back into memory, after it had already been correctly purged. Instead, we just want to resend the reply without loading the peer. (closes issue #16908) Reported by: kkm Patches: 20100412__issue16908.diff.txt uploaded by tilghman (license 14) Tested by: kkm ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e148ffc00f	Also unref the pvt when we delete the provisional keepalive job. (closes issue #16774) Reported by: kowalma Patches: 20100315__issue16774.diff.txt uploaded by tilghman (license 14) Tested by: falves11, jamicque Review: https://reviewboard.asterisk.org/r/591/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	bd53cbabcf	gives channel reference before unlocking it and using setvar helper. To guarantee the channel is valid when calling setvar on the MASTER_CHANNEL dialplan function, a channel reference must be taken before unlocking. Thanks to russell for pointing out the error. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9afa6af881	Remove status_response callbacks where they are not needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	7509949658	Prevent crash when originating a call to a local channel. Call completion code tries to grab the call completion parameters from the requesting channel during local_request. When originating a call to a local channel, however, this channel is NULL. This was causing an issue for me when trying to run a test script. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	537edff10f	Remove PRI CCSS BUGBUG message and update configure script. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	4b8f1c8cac	Add routines for parsing SIP URIs consistently. From the original issue report opened by Nick Lewis: Many sip headers in many sip methods contain the ABNF structure name-andor-addr = name-addr / addr-spec Examples include the to-header, from-header, contact-header, replyto-header At the moment chan_sip.c makes various different attempts to parse this name-andor-addr structure for each header type and for each sip method with sometimes limited degrees of success. I recommend that this name-andor-addr structure be parsed by a dedicated function and that it be used irrespective of the specific method or header that contains the name-andor-addr structure Nick has also included unit tests for verifying these routines as well, so...heck yeah. (closes issue #16708) Reported by: Nick_Lewis Patches: reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis (license 657 Review: https://reviewboard.asterisk.org/r/549 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ae7b76a1b9	Fix some compiler errors that popped up after the CCSS merge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6cad0f1602	func_srv and explicit specification of a remote IP for SIP. From Review Board: There are two interrelated changes here. First, there is the introduction of func_srv. This adds two new read-only dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV records instead. In order to facilitate this work, I added a couple of new API calls to srv.h. ast_srv_get_record_count tells the number of records returned by an SRV lookup. This number is calculated at the time of the SRV lookup. ast_srv_get_nth_record allows one to get a numbered SRV record. Second, there is the modification to chan_sip that allows one to specify a hostname or IP address (along with a port) to send an outgoing INVITE to when dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV records and then use the host and port from the results to dial via a specific host instead of what is configured in sip.conf. Review: https://reviewboard.asterisk.org/r/608 SWP-1200 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d3f44c979c	fixes deadlock in chan_sip caused by usage of MASTER_CHANNEL dialplan function (closes issue #16767) Reported by: lmsteffan Patches: deadlock_16767v3.diff uploaded by dvossel (license 671) Review: https://reviewboard.asterisk.org/r/606/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	1a2f96573a	Merged revisions 256225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256225 \| rmudgett \| 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) \| 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock. SWP-1231 ABE-2163 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256265 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
Russell Bryant	df3fc304c9	Merged revisions 256014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256014 \| russell \| 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) \| 9 lines Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel() (closes issue #16840) Reported by: bzing2 Patches: patch.txt uploaded by bzing2 (license 902) issue_16840.rev1.diff uploaded by russell (license 2) Tested by: bzing2, russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	f4617ae2b5	Ignore Redial softkey when no previous dialed number is known (closes issue #17126) Reported by: wedhorn Patches: skinny79xx_redial1.diff uploaded by wedhorn (license 30) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	cc89bc0a27	Cleanup transmit_* functions Bulk lot of generally trivial changes for cleaning up the transmit stuff. Line state request has been modified for line only responses. (closes issue #16994) Reported by: wedhorn Patches: skinny-clean07.diff uploaded by wedhorn (license 30) Tested by: wedhorn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	284ac3a4d9	Fix improper comaparison of anonymous URI when getting P-Asserted-Identity. There was a bug where we split the URI on the @ sign and then attempted to compare to "anonymous@anonymous.invalid" afterwards. This comparison could never evaluate true. So now we keep a copy of the URI prior to the split so that the comparison is valid. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

1 2 3 4 5 ...

6488 Commits (fa8e44f232bc6fe1c3f2ab504865a43d2f21cf2b)