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Author	SHA1	Message	Date
Sean Bright	b1cf1879bc	Don't leak references if we can't create a pseudo channel for mixing in MeetMe. ... If there was a problem allocating a pseudo channel when building our meetme, we weren't destroying our user container or destroying the mutexes that we created. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	c6985c71da	Merged revisions 304464 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) | 9 lines Fix default prefix=/usr regression on non-Linux systems. This partially reverts a change made in branches/1.4/ r267759, which will cause issue #17013 to be reopened. This issue was pointed out by a user on #asterisk, who helpfully discovered that paths were being set incorrectly. To truly understand what was wrong, one should run: svn diff --force -c<this revision> configure ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	7a339ed6ac	Merged revisions 304460 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan 2011) | 1 line Rerun bootstrap.sh with no changes, so that it is more obvious what my next commit changes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304461 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	7fde40240f	Change delimiter used internally for GOTO_ON_BLINDXFR to commas to match 76703. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	77919e528e	Merged revisions 304242 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed, 26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in ast_udptl ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	051e5a9edf	Merged revisions 304247 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304247 | mnicholson | 2011-01-26 15:00:15 -0600 (Wed, 26 Jan 2011) | 2 lines Convert from network to host byte ordering before checking if an IP is a multicast address. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	afaf018322	Merged revisions 304241 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) | 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	632b2a4417	Merged revisions 304159 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed, 26 Jan 2011) | 1 line Make sure the sample queues.conf is properly commented. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	48e5b9d57c	Merged revisions 304148 from ... https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, 26 Jan 2011) | 2 lines Update documentation for DAHDISendCallreroutingFacility() application. .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	3ef4c6820e	Per the man page, setvbuf() must be called before any other operation on an open file. ... We use setvbuf() to associate a buffer with a stream, but we have already written to the open file. This works (by chance) on Linux, but fails on other platforms, such as OpenSolaris. (closes issue #16610) Reported by: bklang Patches: setvbuf.patch uploaded by crjw (license 963) Tested by: bklang, asgaroth, efutch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	418ce98827	Merged revisions 304005 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011) | 8 lines DTMF attended transfers sometimes fail for no apparent reason. The loop in feature_request_and_dial() can exit when Party C has answered without processing an AST_CONTROL_ANSWER. Also sometimes an AST_CONTROL_ANSWER never happens even though Party C has answered. Don't hangup Party C if he is up or we receive an AST_CONTROL_ANSWER. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	11110b1ead	Merged revisions 303906 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines Guard against retransmitting BYEs indefinitely In the case of an attended transfer (A calls B, A atxfers to C) where A becomes unreachable before replying to Asterisk's BYE, Asterisk can sometimes retransmit the BYE indefinitely. This is because __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out, it will not ever be marked as ALREADYGONE, so when __sip_autodestruct is called again, we end up starting the cycle over. This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt in the case of a BYE that has timed out. This should prevent Asterisk from trying to transmit new BYE messages in the future. Review: https://reviewboard.asterisk.org/r/1077/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0585c53d3f	Fix "sip show user <tab>", so that it actually shows results, instead of just completing the last entry. ... (closes issue #16675) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	569f4c6604	Merged revisions 303765 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	5369a22c18	Merged revisions 303676 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines Fix voicemail sequencing for file based storage. A previous change was made to account for when the number of voicemail messages exceeds the max limit to be handled properly, but it caused gaps in the messages to not be properly handled. This has now been resolved. In later non 1.4 branches, it appears that resequencing wasn't even occurring due from what appears and accidental code removal. (closes issue #18498) Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license 325) (closes issue #18486) Reported by: bluefox Patches: bug18486.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0f8989a993	Merged revisions 303546 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	6d24696968	Merged revisions 303284 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines Reset configuration before parsing users.conf. Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	01dfc39eb4	Fix changes to L() flag in Dial(). ... Tony Mountifield pointed out an error I had in my patch. I was a bit too aggressive on changing 'seconds' to 'milliseconds'. So I decided to do some additioanl testing and have no changed just the appropriate lines. One line says milliseconds, and the other says seconds. Probably should change this to be either just seconds or milliseconds, but I've spent too much time on this already :) (issue #18264) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Shaun Ruffell	8d40dc2122	main/features: Use POLLPRI when waiting for events on parked channels. ... This change resolves a regression in the 1.6.2 when converting from select to poll. The DAHDI timers use POLLPRI to indicate that the timer fired, but features was not waiting for that flag. The result was no audio for MOH when a call was parked and res_timing_dahdi was in use. This patch is slightly modified from the one on the mantis issue. It does not set an exception on the channel if the POLLPRI flag is set. (closes issue #18262) Reported by: francesco_r Patches: patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029) Tested by: francesco_r, rfrantik, one47 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	347d4b4bdb	Merged revisions 303007 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines Add new queue strategy to preserve behavior for when queue members moved to ao2. Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	9aee0b5e0b	Resolve a compiler warning. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	094efca065	Option L() is milliseconds, not seconds. ... > Change the verbose output of option L() to say milliseconds and not seconds > as the value is in milliseconds. > > (closes issue #18264) > Reported by: jacco > Patches: > app_dial_patch.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	39a167a12a	Support greetingsfolder as documented in voicemail.conf.sample. ... (closes issue #17870) Reported by: edhorton Patches: __20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	7289113d93	Turn a noisy verbose message into a debug message. ... This can drown your console if you're using the AMI over HTTP. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	1ce0e49b02	Merged revisions 302671 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines DTMF transfer plays the wrong sounds for wrong number or other call failure. * Set the default for features.conf.sample xferfailsound option to "beeperr" as documented instead of "pbx-invalid" and corrected the use of it in DTMF blind transfer (#1). * Improved DTMF blind transfer handling of wrong numbers. Most of the concerns in this issue were taken care of by the patch for issue 17999: Issues with DTMF triggered attended transfers. (closes issue #18379) Reported by: gincantalupo Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5d7f358fc2	Merged revisions 302663 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19 Jan 2011) | 2 lines Add some API documentation ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	676b6cf22f	Kill zombies. ... When we ast_safe_fork() with a non-zero argument, we're expected to reap our own zombies. On a zero argument, however, the zombies are only reaped when there aren't any non-zero forked children alive. At other times, we accumulate zombies. This code is forward ported from res_agi in 1.4, so that forked children are always reaped, thus preventing an accumulation of zombie processes. (closes issue #18515) Reported by: ernied Patches: 20101221__issue18515.diff.txt uploaded by tilghman (license 14) Tested by: ernied git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	7d8cdc8c05	Don't call strlen() when we only need to look at the next character or two. ... (closes issue #18042) Reported by: wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	faaebdfa2b	Remove an extraneous \r\n at the end of a parking manager events. ... (closes issue #18363) Reported by: clegall_proformatique Patches: asterisk_1.8_295998_parking_manager_events_format.patch uploaded by clegall proformatique (license 1139) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	aa527d8ab2	Properly handle partial reads from fgets() when handling AGIs. ... When fgets() failed with EAGAIN, we were continually decrementing the available space left in our buffer, resulting in botched command handling. (closes issue #16032) Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by fnordian (license 110) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	24fbdc445b	Make sure that h_length is set when we short-circuit out of ast_gethostbyname. ... (closes issue #16135) Reported by: thedavidfactor Patches: utils.patch uploaded by thedavidfactor (license 903) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	c9c8b434f0	Handle 'Resource temporarily unavailable' error more gracefully. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302461 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	ed7b735466	Remove references to priorityjumping from the sample extensions.conf. ... Priority jumping was removed from pbx_config in r68970. (closes issue #18622) Reported by: kshumard Patches: extensions.conf.sample.patch uploaded by kshumard (license 92) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	28cbf8801c	Merged revisions 302311 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) | 4 lines URI encode the user part of the contact header. ABE-2705 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	7ca6a0e2bd	Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip. ... Lock scenario presented here: Thread 1 holds ast_rdlock_contexts &conlock holds handle_statechange hints holds handle_statechange hint waiting for cb_extensionstate Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds handle_request_do &netlock holds find_call sip_pvt_ptr waiting for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911 (ast_rdlock_contexts) Chan_sip has an established locking order of locking the sip_pvt and then getting the context lock. So the as stated by the summary, the operations in thread 2 have been modified to no longer require the context lock. (closes issue #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch uploaded by one47 (license 23), modified by me Review: https://reviewboard.asterisk.org/r/1072/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	f78288c30b	Merged revisions 302172 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines Issues with DTMF triggered attended transfers. Issue #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in features.conf for attended transfer). 3) A hears MOH. B dial number C 4) C ringing. A hears MOH. 5) B hangup. A still hears MOH. C ringing. 6) A hangup. C still ringing until "atxfernoanswertimeout" expires. For v1.4 C will ring forever until C answers the dead line. (Issue #17096) Problem: When A and B hangup, C is still ringing. Issue #18395 SIP call limit of B is 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C ringing 4. Timeout waiting for C to answer 5. Recall to B fails because B has reached its call limit. Because B reached its call limit, it cannot do anything until the transfer it started completes. Issue #17273 Same scenario as issue 18395 but party B is an FXS port. Party B cannot do anything until the transfer it started completes. If B goes back off hook before C answers, B hears ringback instead of the expected dialtone. ********** Note for the issue #17273 and #18395 fix: DTMF attended transfer works within the channel bridge. Unfortunately, when either party A or B in the channel bridge hangs up, that channel is not completely hung up until the transfer completes. This is a real problem depending upon the channel technology involved. For chan_dahdi, the channel is crippled until the hangup is complete. Either the channel is not useable (analog) or the protocol disconnect messages are held up (PRI/BRI/SS7) and the media is not released. For chan_sip, a call limit of one is going to block that endpoint from any further calls until the hangup is complete. For party A this is a minor problem. The party A channel will only be in this condition while party B is dialing and when party B and C are conferring. The conversation between party B and C is expected to be a short one. Party B is either asking a question of party C or announcing party A. Also party A does not have much incentive to hangup at this point. For party B this can be a major problem during a blonde transfer. (A blonde transfer is our term for an attended transfer that is converted into a blind transfer. :)) Party B could be the operator. When party B hangs up, he assumes that he is out of the original call entirely. The party B channel will be in this condition while party C is ringing, while attempting to recall party B, and while waiting between call attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will replace the party B channel technology with a NULL channel driver to complete hanging up the party B channel technology. The consequences of this code is that the 'h' extension will not be able to access any channel technology specific information like SIP statistics for the call. ATXFER_NULL_TECH is not defined by default. ********** (closes issue #17999) Reported by: iskatel Tested by: rmudgett JIRA SWP-2246 (closes issue #17096) Reported by: gelo Tested by: rmudgett JIRA SWP-1192 (closes issue #18395) Reported by: shihchuan Tested by: rmudgett (closes issue #17273) Reported by: grecco Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1047/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	a16089f243	Merged revisions 293493 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 [^] ........ r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines Only offer codecs both sides support for directmedia When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue 0017403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ [^] ........ Backporting a bugfix that should have been included. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	7781c78f06	Blocked revisions 301869 via svnmerge ... ........ r301869 | lmadsen | 2011-01-14 14:21:00 -0600 (Fri, 14 Jan 2011) | 7 lines Fix issue with cross-compiling failing (closes issue #18301) Reported by: abelbeck Patches: asterisk-1.4-bugid18301.patch.txt uploaded by abelbeck (license 946) Tested by: abelbeck, russellb ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Andrew Latham	d4a9e2c20a	Add relationships to function documentation. ... Fix amatuer type mistake git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Andrew Latham	62c5b3d9d4	Add relationships to function documentation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	1566c39525	Add static entry for split Polycom 332 firmware. ... (closes issue #18607) Reported by: cjacobsen Patches: polycom_331.diff uploaded by cjacobsen (license 1029) Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	b381052ca0	Don't reject all SUBSCRIBE auth requests ... When merging another SUBSCRIBE fix from 1.4, some braces were put in the wrong place. This patch fixes that. (closes issue #18597) Reported by: thsgmbh git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	6adc202b2c	Removed a usleep(1) that shouldn't be necessary in session_do, and removed the ... ms_t member from the mansession_session structure. Merged revisions 301591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan 2011) | 5 lines Don't store the thread id for the manager session in the structure we pass to the thread for the manager session. ABE-2543 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	ac9c6a3692	Merged revisions 301502 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) | 12 lines Fix CPU spike when pressing DTMF after agent login. The problem here is that DTMF was being continuously deferred and requeued since ast_safe_sleep is called in a loop. There are serveral other places in the code that sleeps and then loops in a similar fashion. Because of this fact I opted to not defer DTMF any more, which will not affect the original fix: https://reviewboard.asterisk.org/r/674 (closes issue #18130) Reported by: rgj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	44abf69388	Fix a logic issue when passing context ARG ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	423fa1a875	Merged revisions 301305 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan 2011) | 4 lines Prevent buffer overflows in ast_uri_encode() ABE-2705 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	b198610c96	SOUND_CACHE_DIR now defaults to empty ... Sounds files included in the Asterisk tarball were being ignored and re-downloaded. Users wanting to cache the files can still override the setting using the --with-sounds-cache option. (closes issue #18589) Reported by: pabelanger Patches: issue18589.patch uploaded by pabelanger (license 224) Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/1074/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	5e7a60d065	Indicate log level argument for Log() is not optional ... (closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	15bf424dcb	Initialize useropts/adminopts in case there is no column in the realtime DB. ... (closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	1a3a5b1694	Fix regression causing forwarding voicemails to not work with file storage. ... I had actually already fixed this in 295200 in 1.4 and thought it wasn't missing in the other branches for some reason. (closes issue #18358) Reported by: cabal95 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@301046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago