209 Commits (3078814e204af51f7b79b944bf681cbf290774e9)

Author	SHA1	Message	Date
Joshua Colp	3078814e20	Merged revisions 195096 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) | 12 lines Merged revisions 195095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5 lines Fix a bug where the codecs of the called party leg were not properly sent back to the caller call leg when reinvited. (closes issue #13569) Reported by: bkw918 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@195099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	eebe545917	Merged revisions 194209 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r194209 | file | 2009-05-13 10:39:10 -0300 (Wed, 13 May 2009) | 18 lines Merged revisions 194208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) | 11 lines Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. (closes issue #14815) Reported by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue #14460) Reported by: moliveras Tested by: moliveras ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@194213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	8fff8ab669	Merged revisions 188413 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 | file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix an incorrect clock rate when sending T140 text. (closes issue #14029) Reported by: epicac ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@188416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	2f24689b49	Merged revisions 180372 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	d2fb14e26c	Merged revisions 178373 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly. (issue #14460) Reported by: moliveras Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	5c178fb42b	Merged revisions 178141 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines Fix infinite DTMF when a BEGIN is received without an END. This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	3dcdaa5d05	suppress smoothers for Siren codecs as well as Speex and G.723.1 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	2a53f2ec98	Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14) ... This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	12f02a8c11	Merged revisions 175124 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	2bf6d8955d	Merged revisions 170239 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7 lines Don't crash if RTCP is not enabled on an RTP structure but statistics are output. (closes issue #14234) Reported by: jcovert Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551) rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	549fcd78a1	Merged revisions 165591 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us. (closes issue #13545) Reported by: davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	402bd762c0	Merged revisions 162653 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change. (closes issue #12983) Reported by: vt Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	90f6a8eeee	Merged revisions 162204 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment. (closes issue #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file (license 11) Tested by: ip-rob, bujones ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	f02e8e9ea9	Merged revisions 162188 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines Take video into account when early bridging RTP. (closes issue #13535) Reported by: davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	8dd432ec5e	Merged revisions 161013 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines (closes issue #13835) Reported by: matt_b Tested by: jpeeler This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure. Closes AST-142. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	86f900b201	This commit does two things: ... - Add CLI aliases module to asterisk. - Remove all deprecated CLI commands from the code Initial work done by file. Junk-Y and lmadsen did a lot of work and testing to get the list of deprecated commands into the configuration file. Deprecated CLI commands are now handled by this new module, see cli_aliases.conf for more info about that. ok russellb@ via reviewboard (closes issue #13735) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	fd6ee6e1f2	Merged revisions 154060 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) | 3 lines Remove the potential for a division by zero error. (Closes issue #13810) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	e235a07376	(closes issue #13557) ... Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	224980b07b	Merged revisions 143337 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep 2008) | 6 lines Allow for "G.729" if offered in an SDP even though it is not RFC 3551 compliant. Some Cisco switches will send this in an SDP, and it doesn't hurt to be able to accept this. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	08af5bb312	Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating ... when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	9ff1434641	Add missing colons to RTCPReceived and RTCPSent manager events. ... (closes issue #13319) Reported by: srt Patches: 13319_rtcp_manager_event_headers.diff uploaded by srt (license 378) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	790fde68d9	Another batch of files from RSW. The remaining apps and a few more ... files from main/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b69c8e6ab5	Another big chunk of changes from the RSW branch. Bunch of stuff from main/ ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b3970abc30	Merged revisions 136062 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame type, there are places where ast_rtp_new_source may be called where the tech_pvt of a channel may not yet have an rtp structure allocated. This caused a crash in chan_skinny, which was fixed earlier, but now the same crash has been reported against chan_h323 as well. It seems that the best solution is to modify ast_rtp_new_source to not attempt to set the marker bit if the rtp structure passed in is NULL. This change to ast_rtp_new_source also allows the removal of what is now a redundant pointer check from chan_skinny. (closes issue #13247) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	cd16dca459	Merged revisions 129436 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines Fix a problem where inbound rfc2833 audio would be sent to the core instead of being P2P bridged. When the core regenerated the rfc2833 packet for the outbound leg, the SSRC would be different than the RTP audio on the call leg causing DTMF detection issues on the far end. (closes issue #12955) Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by tsearle (license 373) Tested by: tonyredstone ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brett Bryant	d185405755	Janitor project to convert sizeof to ARRAY_LEN macro. ... (closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	945d7022c2	Make this actually evaluate how it was intended to be. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	0a52297cf0	Add new SIP cli command "sip show channelstats" that displays some QoS data (if we have RTCP reports ... and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which would be nice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	84c119cb83	Merged revisions 125276 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008) | 7 lines Check for rtcp structure before trying to delete schedule. (closes issue #12872) Reported by: destiny6628 Patches: 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14) Tested by: destiny6628 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brett Bryant	c1451b5537	This patch adds more detailed statistics for RTP channels, and provides an API call to access it, including maximums, minimums, standard deviatinos, ... and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function for any channel that uses RTP. (closes issue #10590) Reported by: gasparz Patches: chan_sip_c.diff uploaded by gasparz (license 219) rtp_c.diff uploaded by gasparz (license 219) rtp_h.diff uploaded by gasparz (license 219) audioqos-trunk.diff uploaded by snuffy (license 35) rtpqos-trunk-r119891.diff uploaded by sergee (license 138) Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1e9371da8	- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause ... - make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	08f91c1192	Merged revisions 116463 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines Add ast_assert(), which can be used to handle fatal errors. It is only compiled in if dev-mode is enabled, and only aborts if DO_CRASH is defined. (inspired by issue #12650) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	bb386c84e7	Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss in text stream ... Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	5fff9c7304	Merged revisions 114100 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4 lines Don't change the SSRC when a new source comes into play, this might happen quite often and depending on the remote side... they might not like this. (closes issue #12353) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4a21c5dd22	Fix spelling of existent in a few places. ... (closes issue #12409) Reported by: candlerb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	0d7cfae6b6	Merged revisions 112209 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 lines Disable Packet2Packet bridging when we need to feed DTMF frames into the core. Some implementations do not like how we switch between things. (closes issue #12212) Reported by: bamby ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ef4eff9a9b	Add the "config reload <conffile>" command, which allows you to tell Asterisk ... to reload any file that references a given configuration file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	3e439e9616	Merged revisions 110019 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames. (closes issue #11429) Reported by: sperreault Patches: 11429-frametype.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	10cdbe28a8	Merged revisions 109386 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value. (AST-2008-002) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a60f591c72	Merged revisions 106606 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008) | 3 lines Properly initialize rtp->schedid (Closes issue #12154) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5ca5d97673	Merge changes from team/russell/g722-sillyness ... ... Fix a number of other places where the number of samples in a G722 frame was not properly handled because of various reasons. main/rtp.c: - When a G722 frame is read from the smoother, the number of samples in the frame must be divided by 2 before being sent out over the network. Even though G722 is 16 kHz, an error in some previous spec has made it so that we have to list the number of samples such as if it was 8 kHz. main/file.c: - When scheduling the next time to expect a frame, take into account that the format of the file we're reading from may not be 8 kHz. codecs/codec_g722.c: - When converting from G722 to slinear, g722_decode() expects its samples parameter to be in the silly (real samples / 2) format. Make it so. - When converting from slinear to G722, properly set the number of samples in the frame to be the number of bytes of output * 2. formats/format_pcm.c: - This format module handles G722, among a number of other formats. However, the read() and seek() functions did not account for the fact that G722 has 2 samples per byte. (closes issue #12130, reported by rickross, patched by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	496adc6fc0	Merged revisions 106235 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	a760a033e9	Merged revisions 105932 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines Fix a bug that I just noticed in the RTP code. The calculation for setting the len field in an ast_frame of audio was wrong when G.722 is in use. The len field represents the number of ms of audio that the frame contains. It would have set the value to be twice what it should be. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	cfc1df4c1a	Whitespace changes only ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	3b070a815d	Merged revisions 105676 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2 lines In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4de0d8368f	Merged revisions 105674 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines When a new source of audio comes in (such as music on hold) make sure the marker bit gets set. (closes issue #10355) Reported by: wdecarne Patches: 10355.diff uploaded by file (license 11) (closes issue #11491) Reported by: kanderson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	dca12f4aa7	Fix T38 passthrough regression introduced by state changes. ... (closes issue #12078) Reported by: dimas Patches: v1-12078.patch uploaded by dimas (license 88) (closes issue #12074) Reported by: Ivan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	2c3c489ade	Merged revisions 103780 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008) | 9 lines When a SIP channel is being auto-destroyed, it's possible for it to still be in bridge code. When that happens, we crash. Delay the RTP destruction until the bridge is ended. (closes issue #11960) Reported by: norman Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14) Tested by: norman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	c81350d6f6	Just some minor coding style cleanup... ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	1ec8cb41a8	Merge changes from team/mvanbaak/cli-command-audit ... (closes issue #8925) About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI commands in Asterisk 1.4 for the next version of their book, they documented a lot of inconsistencies. This set of changes addresses all of these issues and has been reviewed by Leif. While this does introduce even more changes to the CLI command structure, it makes everything consistent, which is the most important thing. Thanks to all that helped with this one! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago