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2011-07-21Merged revisions 329257 via svnmerge from russell2-2/+2
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines s/1.10/10.0/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@329258 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-19Merged revisions 328824 via svnmerge from kmoore1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines Merged revisions 328823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines RTP bridge away with inband DTMF and feature detection When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Asterisk to allow the call to be locally or remotely bridged, preventing access to the data required to detect activations of such features. (closes 17237) Review: https://reviewboard.asterisk.org/r/1302/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328825 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-19Merged revisions 328717 via svnmerge from twilson1-2/+7
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328717 | twilson | 2011-07-18 20:55:32 -0500 (Mon, 18 Jul 2011) | 14 lines Merged revisions 328716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) | 7 lines Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't modify the element passed in if it isn't found. This commit also adds linked list unit tests. Review: https://reviewboard.asterisk.org/r/1321/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328718 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-15Merged revisions 328329 via svnmerge from rmudgett2-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines Make hint watcher callback take const strings for context and exten parameters. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328344 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-12do v21 detection instead of CED detection for the fax gatewaymnicholson1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327769 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-11Merged revisions 327682 via svnmerge from twilson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) | 9 lines Update chan_gtalk to work with changed GMail-based calls The messages sent by the GMail client have changed, but include the old-style messages as well. This patch checks for this case and uses the old-style offer. (closes issue ASTERISK-18084) Review: https://reviewboard.asterisk.org/r/1312/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327683 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-11Updates follow_talker video_mode in confbridge application.dvossel1-0/+13
follow_talker mode originally echoed the same video stream to all participants. As the primary talker switched around, the video stream would result in the talker seeing themselves. Now the primary talker sees the last person who was talking rather than themselves. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327640 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07I think reviewboard broke this. The whole file was doubled.qwell1-37/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326943 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Adds missing celt.h file from celt pass-through support patch.dvossel1-0/+74
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326900 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Adds pass-through support for codec CELT.dvossel1-0/+2
This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326855 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Video support for ConfBridge.dvossel2-0/+66
Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325931 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Fax gateway functionality (i.e. translating between a T.30 terminal and a T.38mnicholson1-0/+7
terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the dialplan. Big thanks to irroot for porting this code to use the framehooks api. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325816 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-24Merged revisions 324768 via svnmerge from jrose1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) | 11 lines DTMF wasn't being logged on connected consoles when enabled in logger.conf Previously in order for DTMF to be logged in a connected console session, the user would have to do logger set channel DTMF on. This corrects that so that it is on by default. This issue was caused by an off by one error incurred by a logger level count of 6 in logger.h where it should have been 7. (closes issue: ASTERISK-17974) Reported by: Luke H ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324769 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-23Merged revisions 324652 via svnmerge from dvossel1-25/+41
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324652 | dvossel | 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) | 20 lines Merged revisions 324634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver Thanks to twilson for identifying the issue and providing the patches. AST-2011-010 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324664 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-22Merged revisions 324484 via svnmerge from twilson1-2/+73
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines Stop sending IPv6 link-local scope-ids in SIP messages The idea behind the patch listed below was used, but in a more targeted manner. There are now address stringification functions for addresses that are meant to be sent to a remote party. Link-local scope-ids only make sense on the machine from which they originate and so are stripped in the new functions. There is also a host sanitization function added to chan_sip which is used for when peer and dialog tohost fields or sip_registry hostnames are used to craft a SIP message. Also added are some basic unit tests for netsock2 address parsing. (closes issue ASTERISK-17711) Reported by: ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324487 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-21Merged revisions 324364 via svnmerge from dvossel1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) | 10 lines Fixes locking inversion issue in ast_async_goto() During this function we can not hold the "chan" lock while doing the masquerade, the explicit goto on the tmp chan, or the channel alloc. Instead we need to get the channel lock, store off information about the channel that we need, and then let the channel lock go for the remainder of the function. Review: https://reviewboard.asterisk.org/r/1275/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324365 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-16Merged revisions 324048 via svnmerge from twilson1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines Lock the channel before calling the setoption callback The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324050 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Merged revisions 323863 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323863 | twilson | 2011-06-15 14:58:18 -0500 (Wed, 15 Jun 2011) | 2 lines Make ARRAY_LEN() return the same type on x86 and x86_64 systems ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323864 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-14Merged revisions 323370 via svnmerge from twilson1-0/+50
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines Add rtpkeepalives back to 1.8 The RTP-engine conversion left out support for handling rtpkeepalives. This patch adds them back. (closes issue ASTERISK-17304) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323374 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-09Merged revisions 322865 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322865 | twilson | 2011-06-09 15:29:20 -0700 (Thu, 09 Jun 2011) | 4 lines Correct ast_db_deltree documentation ast_db_deltree returns -1 on error, otherwise the number of deletions ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322866 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-09Merged revisions 322749 via svnmerge from rmudgett1-2/+18
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines Remove potential deadlock in call pickup race. Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322750 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-07Correct some whitespace and a reference debug message.rmudgett1-13/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322284 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-06Merged revisions 322069 via svnmerge from jrose1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322069 | jrose | 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) | 8 lines Fixes level toggling for logger set levels since it was reversed (closes issue ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H Review: https://reviewboard.asterisk.org/r/1244/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322070 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-03Merged revisions 321812-321813 via svnmerge from rmudgett1-3/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) | 1 line Correct IAX2 and SIP event subscription description string. ........ r321813 | rmudgett | 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line Constify subscription description parameter string. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321814 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-01Support routing text messages outside of a call.russell4-0/+255
Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321546 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-31Merged revisions 321517 via svnmerge from rmudgett2-3/+5
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321517 | rmudgett | 2011-05-31 15:54:35 -0500 (Tue, 31 May 2011) | 1 line Update some comments. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321518 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-26Merged revisions 321044 via svnmerge from rmudgett1-5/+15
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321044 | rmudgett | 2011-05-26 13:10:17 -0500 (Thu, 26 May 2011) | 1 line Update ast_sockaddr comment with an important note. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321045 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-26Use va_copy for stringfieldstwilson1-5/+5
The ast_string_field_build_va functions were written to take to separate va_lists to work around FreeBSD 4 not having va_copy defined. In the end, we don't support anything using gcc < 3 anyway because we use va_copy all over the place anyway. This patch just simplifies things by removing the second va_list function arguments in favor of va_copy. Review: https://reviewboard.asterisk.org/r/1233/ --This line, and those below, will be ignored-- M include/asterisk/stringfields.h M main/utils.c M main/channel.c git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320946 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-25Merged revisions 320796 via svnmerge from rmudgett1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320796 | rmudgett | 2011-05-25 11:23:11 -0500 (Wed, 25 May 2011) | 17 lines Give zombies a safe channel driver to use. Recent crashes from zombie channels suggests that they need a safe home to goto. When a masquerade happens, the physical part of the zombie channel is hungup. The hangup normally sets the channel private pointer to NULL. If someone then blindly does a callback to the channel driver, a crash is likely because the private pointer is NULL. The masquerade now sets the channel technology of zombie channels to the kill channel driver. Related to the following issues: (issue #19116) (issue #19310) Review: https://reviewboard.asterisk.org/r/1224/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320820 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-23Merged revisions 320560 via svnmerge from kpfleming1-4/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320560 | kpfleming | 2011-05-23 10:47:14 -0500 (Mon, 23 May 2011) | 4 lines Don't generate spurious "No: command not found" messages when running the configure script on a system that has neither gmime-config nor pkg-config. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320561 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-17Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message.rmudgett1-7/+7
The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how much of the current time is put in Q931_IE_TIME_DATE. * Send date/time ie never. * Send date/time ie date only. * Send date/time ie date and hour. * Send date/time ie date, hour, and minute. * Send date/time ie date, hour, minute, and second. * Send date/time ie default: Libpri will send date and hhmm only when in NT PTMP mode to support ISDN phones. (closes issue #19221) Reported by: kenner JIRA SWP-3396 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319427 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-16Merged revisions 319085 via svnmerge from pabelanger1-4/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319085 | pabelanger | 2011-05-16 10:35:21 -0400 (Mon, 16 May 2011) | 10 lines Support gmime-2.4 (closes issue #18863) Reported by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir (license 46) Tested by: tzafrir Review: https://reviewboard.asterisk.org/r/1213/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319086 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-12Merged revisions 318671 via svnmerge from alecdavis1-0/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318672 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-03Merged revisions 316265 via svnmerge from russell1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316293 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-26Merged revisions 315503 via svnmerge from tilghman1-3/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315503 | tilghman | 2011-04-26 14:32:50 -0500 (Tue, 26 Apr 2011) | 28 lines Merged revisions 315502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315502 | tilghman | 2011-04-26 14:22:52 -0500 (Tue, 26 Apr 2011) | 21 lines Merged revisions 315501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines Fix the bounds-checking code. The code that set the bit within the select bitfield was correct, but the bounds-checking code was not. The change to that line uses the new _bitsize macro for clarity. Also, FD_ZERO macro did not zero-out anything but the first word of the bitfield, so this could have caused problems with modules using that macro with the expanded bitfield. (closes issue #18773) Reported by: jamicque Patches: 20110423__issue18773.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@315504 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-21New HD ConfBridge conferencing application.dvossel5-4/+146
Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314598 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-20Introduction of the JITTERBUFFER dialplan function.dvossel2-2/+53
Review: https://reviewboard.asterisk.org/r/1157/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314509 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-20Merged revisions 314417 via svnmerge from rmudgett1-24/+24
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314417 | rmudgett | 2011-04-20 11:54:02 -0500 (Wed, 20 Apr 2011) | 1 line AST_CONTROL_XXX comment changes. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314418 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-18Problems with ISDN MWI to phones.rmudgett1-30/+22
The "controlling user number" is always the number of the voice mail box which is identical with the subscriber number itself. This number which is listed in the ISDN phone MWI menu cannot be called back to contact the voice mail box. The controlling user number should be made configurable. JIRA ABE-2738 JIRA SWP-2846 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314116 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-18Merged revisions 314017 via svnmerge from dvossel1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines sip codec negotiation of dynamic rtp payloads error fix This patch fixes how chan_sip handles dynamic rtp payload types it does not understand. At the moment if a dynamic payload's mime type does not match one we understand, the payload does not get removed from our payload table. As a result of this, the payload is set to whatever dynamic codec we use internally for that payload number on outgoing INVITES. This is incorrect. This patch fixes this by properly checking the rtpmap set function's return code to make sure it was found. The function can return both -1 and -2 depending on the source of the mismatch. We were just checking -1 explicitly. Review: https://reviewboard.asterisk.org/r/1169/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314018 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-11Merged revisions 313279 via svnmerge from lmadsen1-25/+34
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313279 | lmadsen | 2011-04-11 14:36:40 -0500 (Mon, 11 Apr 2011) | 21 lines Merged revisions 313278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313278 | lmadsen | 2011-04-11 14:33:03 -0500 (Mon, 11 Apr 2011) | 14 lines Merged revisions 313277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011) | 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093) Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by tzafrir (license 46) ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@313280 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-01New Feature for chan_dahdi. 4 length pattern matching.jrose1-1/+8
In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s ntax remains the same and the method used to track the pattern history will only change when using the length 4 patterns. (closes issue SWP-3250) Code: jrose rmudgett git-svn-id: http://svn.digium.com/svn/asterisk/trunk@312384 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-01Merged revisions 312286,312288 via svnmerge from tilghman1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312286 | tilghman | 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) | 2 lines Reload must react correctly against a possibly changed table, so dropping the conditional reload flag. ................ r312288 | tilghman | 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) | 21 lines Merged revisions 312287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines Merged revisions 312285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman (license 14) ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@312289 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-31Fix function reference in comment.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@311981 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-11Mix Monitor: Now with r and t options.jrose1-0/+11
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310373 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-07Merged revisions 309808 via svnmerge from tilghman1-4/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines Merged revisions 309251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309809 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-04Add setvar option to calendaringtwilson1-0/+1
Adding the setvar option with variable substitution on the value allows things like setting the outbound caller id name to the summary of a calendar event, etc. Values could be chained together as they are appended in order to do some scripting if necessary. Review: https://reviewboard.asterisk.org/r/1134/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309640 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-28Merged revisions 309035 via svnmerge from tilghman1-4/+11
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines Merged revisions 309033-309034 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error. Detect whether Flex will compile without the workaround; if so, suppress our workaround code. ........ r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify meaning, removing double negative (stupid!) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309036 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵dvossel10-97/+270
audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-15Merged revisions 307879 via svnmerge from rmudgett1-5/+18
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines No response sent for SIP CC subscribe/resubscribe request. Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307883 f38db490-d61c-443f-a65b-d21fe96a405b
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