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2010-04-09Fix some compiler errors that popped up after the CCSS merge.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09Merge Call completion support into trunk.Mark Michelson
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 An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch.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
2010-04-09func_srv and explicit specification of a remote IP for SIP.Mark Michelson
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
2010-04-08Ensure that linker version scripts (used for symbol export control) always ↵Kevin P. Fleming
exist. Using wildcard matching in the Makefile is not adequate to determine whether an export file should exist for a module or not, so instead we'll just create one if the module needs one, or copy the default one if it does not. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-06Mac OS X does not support comparing a mutex to its initializer. Create a ↵Tilghman Lesher
test for this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-06fixes deadlock in chan_sip caused by usage of MASTER_CHANNEL dialplan functionDavid Vossel
(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
2010-04-06Merged revisions 256225 via svnmerge from Richard Mudgett
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
2010-04-05Fix for localchannel.tex to allow PDFs to be generated again.Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.Richard Mudgett
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
2010-04-03Using the Dial application f option when the call is forwarded will likely ↵Richard Mudgett
crash. Fix app_dial.c:do_forward() OPT_FORCECLID setting cid.cid_num with a stack allocated string instead of a heap allocated string. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-02Export MEETMEBOOKID and fix pin-less conferences with realtime conferencesRussell Bryant
(closes issue #16866) Reported by: DEA Patches: rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA Review: https://reviewboard.asterisk.org/r/582/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-02Merged revisions 256014 via svnmerge from Russell Bryant
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
2010-04-02Merged revisions 256009 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010) | 2 lines Remove extremely verbose debug message. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-02Pass the PID of the Asterisk process, not the PID of the canary.Tilghman Lesher
(closes issue #17065) Reported by: globalnetinc Patches: astcanary.patch uploaded by makoto (license 38) Tested by: frawd, globalnetinc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-02Allow symbol export filtering to work properly on platforms that have symbol ↵Kevin P. Fleming
prefixes. Some platforms prefix externally-visible symbols in object files generated from C sources (most commonly, '_' is the prefix). On these platforms, the existing symbol export filtering process ends up suppressing all the symbols that are supposed to be left visible. This patch allows the prefix string to be supplied to the top-level Makefile in the LINKER_SYMBOL_PREFIX variable, and then generates the linker scripts as required to include the prefix supplied. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-02Ignore Redial softkey when no previous dialed number is knownMichiel van Baak
(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
2010-04-02Cleanup transmit_* functionsMichiel van Baak
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
2010-04-01Fix DEBUG_THREADS build on Darwin.Tilghman Lesher
(closes issue #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-01Removed documentation of the non existent 'both' option to 'faxdetect' in ↵Matthew Nicholson
sip.conf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-31Fix improper comaparison of anonymous URI when getting P-Asserted-Identity.Mark Michelson
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
2010-03-31Recorded merge of revisions 255591 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) | 15 lines Ensure line terminators in email are consistent. Fixes an issue with certain Mail Transport Agents, where attachments are not interpreted correctly. (closes issue #16557) Reported by: jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14) Tested by: ebroad, zktech Reviewboard: https://reviewboard.asterisk.org/r/544/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-31Add documentation clarifying when 't' and 'T' can be used.Leif Madsen
(closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-30Merged revisions 255409 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does not start. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-30Merged revisions 255322 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010) | 2 lines Don't make Asterisk not start if pbx_dundi fails to initialize. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-29This patch adds custom device state handling for ConfBridge conferences,Jared Smith
matching the devstate handling of the MeetMe conferences. Review: https://reviewboard.asterisk.org/r/572/ Closes issue #16972 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-29Remove a debugging log entry.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-27corrections in gk interface, small fixes in call clearing.Alexandr Anikin
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-27We need to inclde sys/wait.h on OpenBSD to get WEXITSTATUS.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-27inotify support for pbx_spoolTilghman Lesher
This should give a good speed boost, in that one particular thread isn't waking up once a second to read directory contents. Reviewboard: https://reviewboard.asterisk.org/r/137/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-26Replace some documentation from 1.6.x back into trunk.Leif Madsen
This documentation associated wth tlsbindaddr is still useful so lets synchronize it between trunk and 1.6.x branches. (issue #17054) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-26Update confusing documentation for tlsbindaddr.Leif Madsen
Update some confusing documentation for the tlsbindaddr option in sip.conf.sample. Point at a link instead which has better documentation. (closes issue #17054) Reported by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-26Work around a bug in dash on Ubuntu by checking the number of arguments ↵Sean Bright
before shift'ing. Reported and tested by pabelanger. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Use "local" instead of "system" header file inclusion.Kevin P. Fleming
Now that these files are in the tree, they should prefer the tree's local copy of all Asterisk headers over any that may be installed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Fix a number of other build problems on Mac OS X.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Merged revisions 254800 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) | 1 line Don't remove local copies of utils in uninstall. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Resolve compiler warning on FreeBSD.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Fix chan_ooh323 so it works on Mac OS X, as well.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25chan_usbradio depends on alsa.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Blocked revisions 254714 via svnmergeJason Parker
........ r254714 | qwell | 2010-03-25 14:39:23 -0500 (Thu, 25 Mar 2010) | 4 lines Fix DEBUG_THREADS issue with out-of-tree modules. Take 2, without ABI breakage this time. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Bump cleancount due to ast_channel change.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Remove no-longer-used (and unsafe) field in ast_channel for linked lists.Kevin P. Fleming
The ast_channel structure had a field used for linking a channel into a linked list, but now that ast_channel structures are ao2 objects, this is no longer needed, and could be harmful as ao2 objects really shouldn't ever be placed into linked lists (since those lists don't assist with reference count management on the objects). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Get chan_ooh323 building again after recent build system changes.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Add unit test for testing ACL functionality.Mark Michelson
There are two unit tests contained here. 1. "Invalid ACL" This attempts to read a bunch of badly formatted ACL entries and add them to a host access rule. The goal of this test is to be sure that all invalid entries are rejected as they should be. 2. "ACL" This sets up four ACLs. One is a permit all, one is a deny all, and the other two have specific rules about which subnets are allowed and which are not. Then a set of test addresses is used to determine whether we would allow those addresses to access us when each ACL is applied. This test, by the way, was what resulted in AST-2010-003's creation. Review: https://reviewboard.asterisk.org/r/532 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Merged revisions 254552 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar 2010) | 5 lines Add doxygen for acl.h Review: https://reviewboard.asterisk.org/r/528 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Add new rtpsource options to the CHANNEL function.Mark Michelson
This adds rtpsource options analogous to the rtpdest functions that already exist. In addition, this fixes potential crashes which could result due to trying to read values from nonexistent RTP streams. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Recorded merge of revisions 254452 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection. Here is a copy and paste of the details from my request on reviewboard that dealt with these changes: Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too: seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF. Fix 2. The second change in place is to fix an issue like the following: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end) *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list. Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Merged revisions 254451 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010) | 2 lines Handle new SRCCHANGE control message here too ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Improve handling of T.38 re-INVITEs that arrive before a T.38-capableKevin P. Fleming
application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame *can* fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25handle_speechset has 4 arguments.Leif Madsen
Update code to reflect that handle_speechset has 4 arguments. (closes issue #17093) Reported by: gpatri Patches: res_agi.patch uploaded by gpatri (license 1014) Tested by: pabelanger, mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25remove unneeded explicit channel in dahdi ioctlsTzafrir Cohen
This patch removes some cases where the channel number for an ioctl was passed as a member in a struct rather then through the file descriptor. The gain setting functions passed around a channel which is always 0, and thus this parameter is simply dropped. Review: https://reviewboard.asterisk.org/r/584/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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