asterisk

Commit Graph

Author	SHA1	Message	Date
Alexandr Anikin	71fbf1a748	fix compile error (1.8 don't have ast_channel_name macro) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	1ab70391f8	check rtptimeouts in ooh323 channels as per config file (rtp voice, video, udptl except rtcp) (closes issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches: 19179-ooh323-2.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	ed76cdda72	Replace GNU old-style field designator extensions to fix clang warnings (issue ASTERISK-19540) Reported by: Makoto Dei Patches: clang-gnu-designator.patch uploaded by Makoto Dei (license 5027) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	d58fe85724	Destroy configs when they are no longer used https://reviewboard.asterisk.org/r/1834/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	d2cb0914e4	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	7d0ee59215	call manager_event only if there is not null channel structure (Closes issue ASTERISK-19298) Reported by: robinfood Patches: issue19298.patch uploaded by may213 (License #5415) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355136 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	197662dd4d	Fix outbound DTMF for inband mode (tell asterisk core to generate DTMF sounds). (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	7e7e263652	process null frame pointer returned by ast_rtp_instance_read correctly (closes issue ASTERISK-16697) Reported by: under Patches: segfault.diff (License #5871) patch uploaded by under git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	0c360ed6be	fix issue on channel numbering (calls could have same channel number on heavy loaded system) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	7dc49195d8	Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	8f37bc12c4	move ast_cond_signal for admitted call after all data filled/freed clear all log channels by pointed number not only first free allocated callToken in ooh323_answer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	d4938a111e	Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	c8c5461a76	Fix double alerting, add forced alerting before answer Fix double alerting (it wasn't fixed here by issue #18542) Add forced alerting before connect (if it wasn't before) Try to send all packets from outgoing queue rather than one only Call goes into clearing state when disconnect command is received (closes issue #19361) Reported by: vmikhelson Patches: issue19361-3.patch uploaded by may213 (license 454) Tested by: vmikhelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	ce3d97c483	fix trivial bug in ooh323_indicate on AST_CONTROL_SRC... check p->rtp is not null git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	9b64fbc06c	correct return values in ooh323_indicate for AST_CONTROL_T38_PARAMETERS git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	feb0df1202	Fix branch compile. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	5082c5d44b	Introduce t.38 parameters control functionality not full but enough for Send/RcvFax support Introduce t.38 controls between asterisk core and channel/proto layers. Not all parameters are transferred from proto layers but *Fax apps tested and work ok. (issue #18693) Reported by: benngard2 Patches: issue-18693.patch uploaded by may213 (license 454) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	6d86c8f1e6	added g729onlyA option for announce only AnnexA g.729 codec in h.323 capabilities. Option can be global or per user/peer. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	aa41d4f640	Corrections for properly work with H.323v2 (older) endpoints and other small fixes. Interpret remote side H.225 version. Corrections for H.323v2 endpoints: don't start TCS and MSD before connect, don't start TCS and MSD by accepting H.245 connection, start TCS and MSD by StartH245 facility message. Other fixes: fix non zeroended remoteDisplayName issue, small fixes in call clearing by closing H.245 connection, tcp keepalive introduced on TCP connections (now is hardcoded, will be configurable in the future), don't force H.245tunneling if FastStart is active, don't send Alerting singal more than once per call. (issue 0018542) Reported by: vmikhelson Patches: issue18542-final-3.patch uploaded by may213 (license 454) Tested by: vmikhelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	409feba665	Added fast start and h.245 tunneling options per user and peer. Added options for faststart/h.245 tunneling per user/peer, properly handle these and global options, correction of handling fs/tunneling fields in signalling responses (issue #17972) Reported by: salecha Patches: fs-tunnel-per-point-3.patch uploaded by may213 (license 454) Tested by: may213, salecha git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	ffa69fd54d	Well, who knew chan_ooh323 used udptl? I sure didn't! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 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
Richard Mudgett	9d81fc3273	Fix compile of chan_ooh323.c from IPv6 integration. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274827 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	cd4ebd336f	Add IPv6 to Asterisk. This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	45895dc320	small changes to avoiding 'freeing unused memory...' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	5df6473067	additional checking related to issue 17186 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	91da9be765	Don't pass zero length callerid to ooh323 stack Don't pass zero callerid string to ooh323 stack because it can't encode this properly and can't generate setup message. (closes issue #17186) Reported by: vmikhelson Patches: zero_callerid_num.patch uploaded by may213 (license 454) Tested by: may213 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	89e4c15783	corrections in gk interface, small fixes in call clearing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	fa9d6969d6	generate roundtrip delay requests and responses added response to roundtrip delay requests from opposite side added roundtrip delay request sending to opposite side after answer, added options for sending request (interval between request and count of unreplied requests before forced call hangup) (closes issue #16976) Reported by: vmikhelson Patches: rtdr-1.6.0-2.patch uploaded by may213 (license 454) Tested by: vmikhelson, may213 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	68d1ded8dd	Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	13d7c1e6bd	generate connected line info update from info in h.323 packets Tested by: benngard git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alexandr Anikin	d48fe3c81d	AST_CONTROL_CONNECTED_LINE frame type processing added to setup DisplayIE field incorrect q.931 message order filtered on incoming calls (first msg must be setup, next must be not setup) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242645 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	9c1b53cab4	According to POSIX, the capital L modifier applies only to floating point types. Fixes a crash on Solaris. (closes issue #16572) Reported by: crjw Patches: frame_changes.patch uploaded by crjw (license 963) Plus several others found and fixed by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239074 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alexandr Anikin	4d19e3eb4b	jitterbuffer setup correction correction of double pointer references from previous rev git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232853 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f59fe83c56	More 32->64 bit codec conversions. In the process of swapping ULAW to a place in the extended codec space, we found several unhandled cases, where a 32-bit integer was still being used to handle a codec field. Most of these have been fixed with this commit, although there is at least one case (codec_dahdi) which depends upon outside headers to be altered before a conversion can be made. (Fixes AST-278, SWP-459) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	341c7b60dd	Make compilation of chan_ooh323 disabled by default. All addons modules should be disabled by default, requiring the user to turn them on if desired. After all, these are addons we're talking about here. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	f999554af7	Update chan_ooh323 to support the expanded codec bitfield from 227580. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alexandr Anikin	9530310954	Reworked chan_ooh323 channel module. Many architectural and functional changes. Main changes are threading model chanes (many thread in ooh323 stack instead of one), modifications and improvements in signalling part, additional codecs support (726, speex), t38 mode support. This module tested and used in production environment. (closes issue #15285) Reported by: may213 Tested by: sles, c0w, OrNix Review: https://reviewboard.asterisk.org/r/324/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227898 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
David Brooks	48363c16e1	Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	1ae0291374	Rename ooh323.conf to chan_ooh323.conf, make module support both names git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c511a26749	Move Asterisk-addons modules into the main Asterisk source tree. Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

42 Commits (02fb0d8d28c123a9b998f78a77d0e33d30a80f2c)