asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	bfaa341f58	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/branches/1.4@175124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	70f7c7e9cb	Revert RTP changes for continuation of DTMF. Proxy commit by russell via SMS. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	1d4e4ff3d1	Clear out the current event after forcing the end of a digit git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	9dff8995b4	Fixify infinite DTMF in the case that no RFC2833 END event is ever received git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	9e38dd5427	Restore a behavior that was recently changed, when we fixed issue #13962 and issue #13363 (related to issue #6176). When a hangup occurs during a Macro execution in earlier 1.4, the h extension would execute within the Macro context, whereas it was always supposed to execute only within the main context (where Macro was called). So this fix checks for an "h" extension in the deepest macro context where a hangup occurred; if it exists, that "h" extension executes, otherwise the main context "h" is executed. (closes issue #14122) Reported by: wetwired Patches: 20090210__bug14122.diff.txt uploaded by Corydon76 (license 14) Tested by: andrew git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Matthew Nicholson	b60ec2baa1	Improve behavior of jitterbuffer when maxjitterbuffer is set. This change improves the way the jitterbuffer handles maxjitterbuffer and dramatically reduces the number of frames dropped when maxjitterbuffer is exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all new frames were dropped until the jitterbuffer is empty. This change modifies the code to only drop frames until maxjitterbuffer is no longer exceeded. Also, previously when maxjitterbuffer was exceeded, dropped frames were not tracked causing stats for dropped frames to be incorrect, this change also addresses that problem. (closes issue #14044) Patches: bug14044-1.diff uploaded by mnicholson (license 96) Tested by: mnicholson Review: http://reviewboard.digium.com/r/144/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	4e069885ce	Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	c257ffeed0	Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	13a60eba0c	This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	0b74f727d7	Prevent a crash from occurring when a jitter buffer interpolated frame is removed from a slinfactory slinfactory used the "samples" field of an ast_frame in order to determine the amount of data contained within the frame. In certain cases, such as jitter buffer interpolated frames, the frame would have a non-zero value for "samples" but have NULL "data" This caused a problem when a memcpy call in ast_slinfactory_read would attempt to access invalid memory. The solution in use here is to never feed frames into the slinfactory if they have NULL "data" (closes issue #13116) Reported by: aragon Patches: 13116.diff uploaded by putnopvut (license 60) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@171621 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	87c02936b6	When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them. (closes issue #14249) Reported by: RadicAlish git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@170648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	1bef118f00	Fix broken call pickup There was a subtle change in ast_do_masquerade which resulted in failed attempts to pickup calls. The problem was that the value of the AST_FLAG_OUTGOING flag was copied from the clone to the original channel. In the case of call pickup, this meant that the AST_FLAG_OUTGOING flag ended up being cleared on the channel that was attempting to execute the pickup. Because this flag was not set, when ast_read came across an answer frame, it ignored it. The result of this was that the calling channel was never properly answered. This fix changes the behavior in ast_do_masquerade to set the flags on the original channel to the union of the flags on the clone channel. This way, if the AST_FLAG_OUTGOING flag is set on either of the two channels involved in the masquerade, the resulting channel will have the flag set as well. (closes issue #14206) Reported by: francesco_r Patches: 14206.patch uploaded by putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@170392 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	3ff70ed000	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/branches/1.4@170239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	4ee4e941f8	Do a string comparison instead of pointer comparison since some people specify the context they are actually in as an argument to get around some funkiness. (closes issue #14011) Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga (license 665) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@170050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	376d85f96c	Read lock the contexts to maintain the locking order when we are notified that the state of a device has changed. (closes issue #13839) Reported by: mcallist git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	9db9bae4d9	Adding revision 169794 to 1.4 since 1.4 is also affected by the issue git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	e34da1e519	Extra NULLs in the output cause some terminal types to abort in the middle of a color code, causing terminal weirdness. (closes issue #14130) Reported by: coolmig Patches: 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, coolmig git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	f311539733	Truncate userevents at the end of a line, when the command exceeds the buffer. (closes issue #14278) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	d671bb1404	Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	99313c7b92	Don't read into a buffer without first checking if a value is beyond the end. (closes issue #13600) Reported by: atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76 (license 14) Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	9161b7fc87	Revert unnecessary indications API change from rev 122314 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	5fe8bde41a	Fix the last couple of places where free() was improperly used directly. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@167566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	2839e074ef	Don't fclose() the file early, the filestream destructor will handle it. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@167554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	301945890d	Only try to close the file if one was actually opened git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@167545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c67d152525	Don't use free() directly. This caused a crash since ast_filestream is now an ao2 object. Reported by JunK-Y on IRC, #asterisk-dev git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@167541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	24ccfad6dc	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/branches/1.4@167432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	5eca294b65	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/branches/1.4@167299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	5dd486be22	Compile, even if both DAHDI and Zaptel are not installed. (Closes issue #14120) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@166592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	62c8625f2e	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/branches/1.4@166568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	45bc54db0a	Use the integer form of condition for integer comparisons. (closes issue #14127) Reported by: andrew git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@166509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	8b68bd17e5	Fix up timeout handling in ast_carefulwrite(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@166297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7fdf99803e	Backport of AUDIOHOOK_INHERIT for Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@166157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	e0bec5d67d	(closes issue #13480 ) Reported by: tzafrir Replace a bunch of if defined checks for Zaptel/DAHDI through several new defines in dahdi_compat.h. This removes a lot of code duplication. Example from bug: #ifdef HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd = open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd = open(DAHDI_FILE_PSEUDO, O_RDRW); git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@165991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	991494f2c0	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/branches/1.4@165796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	3a354c3500	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/branches/1.4@165591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	63444acf6b	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/branches/1.4@164881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	aa60ad0e43	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/branches/1.4@164806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	083047a794	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/branches/1.4@164736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	4f807bb183	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/branches/1.4@164634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	3a864cb7b6	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/branches/1.4@164201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	6d268e6d39	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/branches/1.4@163761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c518ed3be1	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/branches/1.4@163448 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	f6ef5d5d6c	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/branches/1.4@163383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	a6edd8ba5f	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/branches/1.4@162653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c2446000d9	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/branches/1.4@162413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	b234c024a0	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/branches/1.4@162265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	114c659195	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/branches/1.4@162204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	7209e0e173	Take video into account when early bridging RTP. (closes issue #13535) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	53c30bd359	Fix a problem with GROUP() settings on a masquerade. The previous code carried over group settings from the old channel to the new one. However, it did nothing with the group settings that were already on the new channel. This patch removes all group settings that already existed on the new channel. I have a more complicated version of this patch which addresses only the most blatant problem with this, which is that a channel can end up with multiple group settings in the same category. However, I could not think of a use case for keeping any of the group settings from the old channel, so I went this route for now. (closes AST-152) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@161948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Sean Bright	ffc0c7e4ae	Merged revisions 161421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r161421 \| seanbright \| 2008-12-05 15:50:23 -0500 (Fri, 05 Dec 2008) \| 8 lines Fix build errors on FreeBSD (uint -> unsigned int). (closes issue #14006) Reported by: alphaque Patches: astobj2.h-patch uploaded by alphaque (license 259) (Slightly modified by seanbright) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@161426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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992 Commits (3e396d458ad5eeeae1b28e10e050ead501bbb039)