43 Commits (ebaf59a656ef94e19332c08e0d0562b8cbe59f65)

Author	SHA1	Message	Date
Terry Wilson	ebaf59a656	Opaquification for ast_format structs in struct ast_channel ... Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types ... Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	34c55e8e7c	Opaquify char * and char[] in ast_channel ... Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions ... There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Nicholson	bb07ca66a1	Merged revisions 340109 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340109 | mnicholson | 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) | 18 lines Merged revisions 340108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0096238b52	Merged revisions 320823 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	3556e4c2d4	Replace ast_log(LOG_DEBUG, ...) with ast_debug() ... (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields ... This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	15 years ago
Matthew Nicholson	f3a9392542	Don't pass null to manager_event() ... (closes issue #17087) Reported by: bklang Patches: app-fax-null-sprintf1.diff uploaded by mnicholson (license 96) Tested by: bklang git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	8ef8706944	Updated CHANGES file to mention res_fax and res_fax_spandsp. ... Also fixed MODULEINFO depends and conflicts for app_fax, res_fax, and res_fax_spandsp. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	2d6f0c4607	Ensure frames are only freed once. ... (closes issue #16361) Reported by: vlad Patches: 20100208__issue16361.diff.txt uploaded by tilghman (license 14) Tested by: kenny, bloodoff, misaksen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	3760672f40	Don't offer MMR or JBIG transcoding during T.38 negotiation. ... After further discussion with Steve Underwood, we should not (yet) be offering to receive MMR or JBIG transcoded streams from T.38 endpoints. A future spandsp release will support those features, and then they can be enabled during negotiation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	df1fc1f381	spandsp does in fact support V.17 modulation at 14.4 kilobits per second, ... so we should generate T38MaxBitRate of 14400 (even though that doesn't really affect the FAX transmission much at all) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	80fc9540f9	Fix another buglet in T.38 session teardown at the end of FAX sessions. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	237795f6d7	Ensure that only one end of a T.38 session initiates teardown at completion. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5e2aa190fe	Display a list of channel variables in each channel-oriented event. ... (Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 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
Kevin P. Fleming	e197f85b8c	Remove automatic switching from T.38 to voice mode in chan_sip. ... chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	7c71e98879	Initiate T.38 switchover when acting as called party, regardless of FAX direction. ... SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	eb449d514e	Minor improvements to app_fax. ... This patch makes some small changes to handle watchdog timeouts in a better way, and also uses a 'cleaner' method of including the spandsp header files. (closes issue #14769) Reported by: andrew Patches: app_fax-20090406.diff uploaded by andrew (license 240) v1-14769.patch uploaded by dimas (license 88) Tested by: freh, deti, caspy, dimas, sgimeno, Dovid git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	7c5f20e46b	Cleanup T.38 negotiation changes. ... Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages, clean up some looping logic, and correct an improper use of ast_free() for freeing an ast_frame. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	b6b3fed0c7	Make T.38 switchover in ReceiveFAX synchronous. ... In receive mode, if the channel that ReceiveFAX is running on supports T.38, we should *always* attempt to switch T.38, rather than listening for an incoming CNG tone and only triggering on that. The channel may be using a low-bitrate codec that distorts the CNG tone, the sending FAX endpoint may not send CNG at all, or there could be a variety of other reasons that we don't detect it, but in all those cases if T.38 is available we certainly want to use it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	0a6e06c7ff	Rework of T.38 negotiation and UDPTL API to address interoperability problems ... Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	44af00a28f	Eliminate extraneous LOG_DEBUG messages generated by app_fax. ... The transmit_audio() and transmit_t38() functions in app_fax have processing loops that are supposed to wait for frames to arrive on the channel and then handle them, but they also have short timeouts so that the loops can have watchdog timers and do other required processing. This commit changes the loops to not actually call ast_read() and attempt to process the returned frame unless a frame actually arrived, eliminating hundreds of LOG_DEBUG messages and slightly improving performance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	c75a129323	Fix some remaining T.38 negotiation problems in app_fax. ... Revision 205696 did not quite fix all the issues with the T.38 negotiation changes and app_fax; this patch corrects them, along with a couple of other minor issues. (closes issue #15480) Reported by: dimas Patches: test2-15480.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	67d1957e60	Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. ... Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	59c1998d67	Improve T.38 negotiation by exchanging session parameters between application and channel. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	2fc4832e85	Fix version detection for API changes in spandsp. ... (closes issue #15355) Reported by: deuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	f1dc620467	Enable applications to enable/disable digit and tone detection. ... Some applications (notably app_fax) do not need digit detection nor FAX tone detection while they are running, and if Asterisk is using software DSPs to provide the detection, this consumes extra CPU cycles that could be better spent on the actual application. This patch allows applications to query and control the state of digit and tone detection on a channel, and modifies app_fax to disable them while the FAX operations are occurring (and re-enable digit detection afterwards). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	aaeec3b40f	Last batch of 'static' qualifiers for module-level global variables. ... Fix up modules in the 'apps' directory, and also correct the bad example of enum definitions in include/asterisk/app.h, which many developers followed (thanks for reading the documentation!). In addition, add some basic usage examples of the 'pahole' and 'pglobal' tools to the coding guidelines. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	12ff77f975	Global var cleanup - constification and removing unused vars. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 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	16 years ago
Dwayne M. Hubbard	83bbf83f49	Make app_fax compatible with spandsp-0.0.6pre4 ... Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred integer to indicate the number of pages transferred (so far) during the fax session. The spandsp-0.0.6pre4 release removed the pages_transferred integer and replaced it with two different integers - pages_tx and pages_rx. This revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards compatibility for previous spandsp releases. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	8be6bc5f67	Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret. ... (closes issue #14073) Reported by: seandarcy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5b168ee34b	Merge changes from team/group/appdocsxml ... This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6d5409e23d	API differences in spandsp 0.0.6pre1 and higher ... (closes issue #13688) Reported by: irroot Patches: app_fax-span6.patch uploaded by irroot (license 52) with minor modifications by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	ccfbd9ab15	Add FAXMODE variable with what fax transport was used. ... (closes issue #13252) Patches: v1-13252.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	2ed6cea551	Update app_fax for better compatibility with spandsp 0.0.5. Add a call to ... t38_terminal_release, and make sure that the phase E handler gets called with proper status. (closes issue #13020) Reported by: dimas Patches: v1-appfax.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	6c49489b00	Resurrected app_fax ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	ef3b214728	Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5b7344ab66	Add app_fax from asterisk-addons, with some additional changes to resolve compiler ... warnings, as well as update to the APIs in spandsp 0.0.5. Spandsp 0.0.5 is being distributed under the LGPL, so we can move this module into the main tree. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago