asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	8ccb6795e9	Merged revisions 167432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167432 \| russell \| 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009) \| 4 lines Treat an empty string the same way as a NULL country argument. In passing, simplify the handling of returning a default tone zone. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	483c555c41	Merged revisions 167299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167299 \| mmichelson \| 2009-01-06 15:35:57 -0600 (Tue, 06 Jan 2009) \| 8 lines Use the correct variable when creating the format string (closes issue #14177) Reported by: nic_bellamy Patches: asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic (license 299) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	334a89f754	When parsing environment variable ASTERISK_PROMPT, make sure to proceed to the next character when a non format specifier is used (no %). Otherwise, the while loop looking for the null byte will never exit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	c70d486ec6	Don't forget to free typename git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	dca80878af	That was weird... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	7cb7920e19	Merged revisions 166953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166953 \| tilghman \| 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) \| 5 lines Also inherit the musiconhold class. (Closes #14153) Reported by: Jerry Geis, via the users list. Patch by: me (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166954 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	aa905e347e	Merged revisions 166093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 \| murf \| 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) \| 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	859ae78977	Merged revisions 166568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166568 \| mmichelson \| 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) \| 12 lines Fix a crash resulting from a datastore with inheritance but no duplicate callback The fix for this is to simply set the newly created datastore's data pointer to NULL if it is inherited but has no duplicate callback. (closes issue #14113) Reported by: francesco_r Patches: 14113.patch uploaded by putnopvut (license 60) Tested by: francesco_r ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	18e07935ed	Merged revisions 166509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166509 \| tilghman \| 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) \| 4 lines Use the integer form of condition for integer comparisons. (closes issue #14127) Reported by: andrew ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	7166156658	Remove some error messages. This is the default handler that is valid to use. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	50387142b7	Merged revisions 166297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166297 \| russell \| 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) \| 2 lines Fix up timeout handling in ast_carefulwrite(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c2999a8366	Introduce ast_careful_fwrite() and use in AMI to prevent partial writes. This patch introduces a function to do careful writes on a file stream which will handle timeouts and partial writes. It is currently used in AMI to address the issue that has been reported. However, there are probably a few other places where this could be used. (closes issue #13546) Reported by: srt Tested by: russell http://reviewboard.digium.com/r/104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	29b658f040	Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port. (closes issue #13628) Reported by: pananix Patches: bug13628.patch uploaded by jpeeler (license 325) Tested by: file, blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	9f7ce9da41	Fix a file playback crash and explicitly initialize values in func_timeout.c A crash was brought up on the bugtracker. The first run through valgrind was full of legitimate complaints of uninitialized values in func_timeout when setting a response timeout. These were fixed but the crash persisted. A second run through showed the real problem. The reference counting used for filestreams was incorrect because there were some missing increments when a frame was read from a format module. (closes issue #14118) Reported by: blitzrage Patches: 14118v2.patch uploaded by putnopvut (license 60) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	e015e6f404	Get rid of an extra space. I don't know how this crept back in when I had already fixed it earlier git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	9733b30ff0	Adding a new dialplan function AUDIOHOOK_INHERIT This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	1cb4baade2	Merged revisions 165796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165796 \| russell \| 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) \| 11 lines Make ast_carefulwrite() be more careful. This patch handles some additional cases that could result in partial writes to the file description. This was done to address complaints about partial writes on AMI. (issue #13546) (more changes needed to address potential problems in 1.6) Reported by: srt Tested by: russell Review: http://reviewboard.digium.com/r/99/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	4e4093ab48	(closes issue #13993 ) Reported by: mika Add ActionID response to ping if sent with request. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	50a25ac847	Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 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	18 years ago
Eliel C. Sardanons	344a37f2a7	Remove duplicate code from the ast_str API. We now use __AST_STR_* to access 'struct ast_str' members, but this must only be used inside the API implementation. (closes issue #14098) Reported by: eliel Patches: ast_str.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	27cbfc1bd5	Add timezone to the possible fields in a timespec. (closes issue #14028) Reported by: mostyn Patches: timezone-v2.patch uploaded by mostyn (license 398) (with additional code guideline fixes and a memory leak fix by me - license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c8da171dd1	Merged revisions 164881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164881 \| russell \| 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008) \| 9 lines Fix an issue where DEBUG_THREADS may erroneously report that a thread is exiting while holding a lock. If the last lock attempt was a trylock, and it failed, it will still be in the list of locks so that it can be reported. (closes issue #13219) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	53f788c6b5	Fix build issues on Linux after sysinfo related changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	0859a4e30c	Merged revisions 164806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164806 \| russell \| 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) \| 9 lines Add "restart gracefully" to the AMI blacklist of CLI commands. "module unload" was already identified as a command that can not be used from the AMI. "restart gracefully" effectively unloads all modules, and will run in to the same problems. (closes issue #13894) Reported by: kernelsensei ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	d2d96b10ac	introduce 'core show sysinfo' for systems that dont have the Linux-ish sysinfo stuff but do have sysctl. (closes issue #13433) Reported by: mvanbaak Patches: 2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license 7) with two free calls replaced with ast_free based on feedback on reviewboard Review: http://reviewboard.digium.com/r/91/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	9de00f16f6	(closes issue #14076 ) Reported by: toc Tested by: murf OK, Well this issue has had its share of flip-flopping. I found the following: 1. the code in question, in ext_cmp1 in pbx.c, would not allow two extensions that vary only by any dashes contained within them, to be defined in the same context. 2. for input dialstrings, dashes are NOT ignored. So, skipping them when sorting patterns seemed a bit silly. Thus, you might declare ext 891 in a context, but if you try dialing 8-9-1, it will NOT match 891. So, I proposed to remove the code from ext_cmp1 to skip the spaces and dashes. Just kept us from declaring 891 and 8-9-1 in the same context, forcing users to generate otherwise uselessly obfuscated dialplan code to get the same effect. Then, I tried out 1.4, and found that: 1. you can declare 891 and 8-9-1 in the same context! 2. You can't define 891, and have 8-9-1 match it! Nor can you define 8-9-1, and have 891 match it! So, it appears that my proposal simply restores the pbx to behaving as it did in 1.4. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	556b082522	Merged revisions 164736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164736 \| russell \| 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) \| 14 lines Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was being used within the context of the thread local data destructors. We would go off and allocate more thread local data while the pthread lib was in the middle of destroying it all. This led to a memory leak. Another issue was an invalid argument being provided to the the object_add API call. (closes issue #13678) Reported by: ys Tested by: Russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164737 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	eb73f5673a	Merged revisions 164634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164634 \| murf \| 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) \| 5 lines I added a sentence to clarify why - and ' ' are ignored in patterns as per bug 14076. Leif says he'll put some stuff about it in the extensions.conf sample, etc. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	5a5cb18f54	Make sure we handle a uint32_t payload in ast_frdup() (closes issue #14080) Reported by: fnordian Patches: frame.patch uploaded by fnordian (license 110) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	d330d3e210	Use ast_seekstream to return the file stream back to the beginning instead of directly seeking to zero. This is because some audio formats have headers at the front that need to be skipped, which will be done by the format module. (closes issue #14079) Reported by: elguero git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	6df30fb8cc	Update to work with new ast_str changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c9eb01c899	Merged revisions 164201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164201 \| russell \| 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) \| 31 lines Handle a case where a call can be bridged to a channel that is still ringing. The issue that was reported was about a case where a RINGING channel got redirected to an extension to pick up a call from parking. Once the parked call got taken out of parking, it heard silence until the other side answered. Ideally, the caller that was parked would get a ringing indication. This patch fixes this case so that the caller receives ringback once it comes out of parking until the other side answers. The fixes are: - Make sure we remember that a channel was an outgoing channel when doing a masquerade. This prevents an erroneous ast_answer() call on the channel, which causes a bogus 200 OK to be sent in the case of SIP. - Add some additional comments to explain related parts of code. - Update the handling of the ast_channel visible_indication field. Storing values that are not stateful is pointless. Control frames that are events or commands should be ignored. - When a bridge first starts, check to see if the peer channel needs to be given ringing indication because the calling side is still ringing. - Rework ast_indicate_data() a bit for the sake of readability. (closes issue #13747) Reported by: davidw Tested by: russell Review: http://reviewboard.digium.com/r/90/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	c8223fc957	Merge ast_str_opaque branch (discontinue usage of ast_str internals) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	5e034d9f0b	Merged revisions 163761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163761 \| tilghman \| 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) \| 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a pointer inside editline to look back to asterisk.c, so others don't spend as much time as I did looking (in the wrong place) for the appropriate function. Reported by: ZX81, via the #asterisk-users channel Fixed by: me (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	90e65dc7d3	Rename a number of tcptls_session variables. There are no functional changes here. The name "ser" was used in a lot of places. However, it is a relic from when the struct was a server_instance, not a session_instance. It was renamed since it represents both a server or client connection. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	035a7552d6	Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven. (closes issue #13525) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	7fcac067b2	Merged revisions 163448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 \| russell \| 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) \| 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	592cab8202	Merged revisions 163383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163383 \| tilghman \| 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) \| 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal is messed up. By intercepting those events with a signal handler in the remote console, we can avoid those issues. (closes issue #13464) Reported by: tzafrir Patches: 20081110__bug13464.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7828e7a966	Add an appropriate goto if ast_call fails git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	fb242bc8fd	Fix the "failed" extension for outgoing calls. The conversion to use ast_check_hangup() everywhere instead of checking the softhangup flag directly introduced this problem. The issue is that ast_check_hangup() checked for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances, such as with a dummy channel. The fix is simple. Don't check tech_pvt. It's pointless, because the code path that sets this to NULL is when the channel hangup callback gets called. This happens inside of ast_hangup(), which is the same function responsible for freeing the channel. Any code calling ast_check_hangup() better not be calling it after that point, and if so, we have a bigger problem at hand. (closes issue #14035) Reported by: erogoza git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	62130ba876	Reduce indentation level of ast_feature_request_and_dial git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	31e068ade2	Merged revisions 163092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163092 \| russell \| 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) \| 11 lines Fix an issue that made it so you could only have a single caller executing a custom feature at a time. This was especially problematic when custom features ran for any appreciable amount of time. The fix turned out to be quite simple. The dynamic features are now stored in a read/write list instead of a list using a mutex. (closes issue #13478) Reported by: neutrino88 Fix suggested by file ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	8c89090160	Previously missing line, now the substitution works correctly git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	689465ba98	Checking global variables here actually overwrote the previous substitution by channel variables, and in any case, was redundant; pbx_substitute_variables_helper ALREADY does substitution for global variables. (closes issue #13327) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	c8c8995b70	add tab completion for 'core set debug X filename.c' (closes issue #13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 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	18 years ago
Russell Bryant	d0bc22b3e8	Add some additional Asterisk project developer documentation. After the nightly update of the documentation on asterisk.org, I'll post an update to asterisk-dev with a pointer to the changes. This covers some release branch and commit policy information. None of this should be a surprise, since it's just documenting what we have already been doing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	179667088b	Merged revisions 162413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162413 \| russell \| 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) \| 8 lines Remove the test_for_thread_safety() function completely. The test is not valid. Besides, if we actually suspected that recursive mutexes were not working, we would get a ton of LOG_ERROR messages when DEBUG_THREADS is turned on. (inspired by a discussion on the asterisk-dev list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	5f4dc23293	Merged revisions 162265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162265 \| mmichelson \| 2008-12-09 14:28:44 -0600 (Tue, 09 Dec 2008) \| 6 lines If we fail to start a thread for the pbx to run in, we need to be sure to decrease the number of active calls on the system. This fix may relate to ABE-1713, but it is not certain yet. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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