asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	3cbc42e2e4	Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	a3769669b0	Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	52752df83c	Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Richard Mudgett	00c1a47c88	Use the correct branch integrated property when generating the version string. Copied the make_version file from Asterisk trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181898 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	8dbfea83ce	Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	1813d4a281	Fix incorrect usage of strncasecmp... I really meant to use strcasecmp. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	01b90d9092	Fix logic flaw in previous commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	2e825bd259	Fix another scenario where depending on configuration the stream would not get read. For custom commands we don't know whether the audio is coming from a stream or not so we are going to have to read the data despite no channels. (closes issue #14416) Reported by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	293900d3f2	Fix issue with streaming MOH failing if nobody is listening. When a music class is setup to actually provide music on hold from a stream we need to constantly read audio from it since it will constantly be providing audio. This is now done despite there being no channels listening to it. (closes issue #14416) Reported by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	e809699406	Allow prefix to set localstatedir (when used and different from the default). This is similar to the /etc change that was made for the non-FreeBSD case. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	aedf566905	Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	f97c929946	encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	563c72dc84	Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	b15b319bd6	Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	21ca773c28	Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. Because using the ast prefix calls are a better choice, ast_free_ptr is the new wrapper for free to pass to functions. Also, a little bit of clean up was done to avoid the debug macros intentionally being redefined. (closes issue #13593) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	280153085e	Remove unused variables. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	849820fd54	Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	df31bb22c0	Make things happier when using autoconf 2.62+ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	a8e2597803	Make compilation succeed in dev-mode when IMAP storage is enabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	04836d554d	Fix handling of backreferences for ENUM lookups enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	aef6c114f1	[IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts. There was a fix put in a while back so that an X-Asterisk-VM-Context message header was added to stored IMAP voicemails. This would allow for us to differentiate if the same mailbox name was used in multiple contexts. The problem still left was that not all places where messages were retrieved actually attempted to use this header for information when retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain work as expected. (closes issue #13853) Reported by: vicks1 Patches: 13853_v2.patch uploaded by mmichelson (license 60) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7e44582f57	Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	5436d8709f	Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	c42b21bc6a	Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion. (issue #AST-194) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	5a3bc6b38d	Make sure we still support zapchan in users.conf, in addition to dahdichan. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	ab5b88843c	Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	3ef0938c76	Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	604a51f341	These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text. I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	2e4471d758	Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	3f2a1247f4	Move where fdno is set to the default value to after the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	a1d249577e	Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	4055ec6c57	Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	dadbbb0a56	Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	b705454875	When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	6706e0be24	Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179461 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	5062dff9dd	Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	c2f0803e77	Update documentation for DIALEDTIME and ANSWEREDTIME variables. (closes issue #14566) Reported by: klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	5996b58192	This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	cbd35b45af	IAX2 prune realtime fix Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	6fb39726c5	This patch prevents the feature detection timeout from being cut in half. Because the ast_channel_bridge() call will return 0 and pass a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer field in hte config struct is getting decremented twice, which effectively cuts the digittimeout in half. I added conditions to the if statement to only let DTMF_END frames to flow thru, which solved the problem. Also, when the frame pointer is null, let control flow thru-- this usually happens on timeouts. I added a comment to the code to explain what's going on and why. Many thanks to sodom for reporting this problem. Personnally, it always seemed like something was wrong with the featuredigittimeout, but I never could quite decide what... and was too busy to investigate. This bug forced the issue, and now we know. Sodom had other issues in 14515, but I couldn't reproduce them. If he still has problems, and wants to get them solved, he is welcome to reopen 14515. (closes issue #14515) Reported by: sodom Patches: 14515.patch uploaded by murf (license 17) Tested by: murf, sodom git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	c4d2dc7967	This patch completes the fixes nec. to make 1.4 asterisk dialplan expressions ($[...]) 8-bit transparent While I was updating ast_expr2.fl, I missed one rule that would allow 8-bit chars to be caught in tokens; and in so doing, it absorbs the ${ sequence and messes up the checking of raw exprs by AEL. Trunk already has these changes. (closes issue #14543) Reported by: klaus3000 Patches: patch.14543 uploaded by murf (license 17) Tested by: murf git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	7374b9195f	Update the copyright year for the main page of the doxygen documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	48933e0c48	Add section about the #exec command in configuration files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	da0d84c3e4	Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly. (issue #14460) Reported by: moliveras Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	cdf5240895	Change include order to make compile on Centos 5 with DAHDI If BIT_TYPES_DEFINED gets defined before linux/types.h is included, the __s32 type doesn't get defined git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	aa488ca6b0	Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	aa5a927b69	Fix infinite DTMF when a BEGIN is received without an END. This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	4af0175285	Don't print the CR-NL combination when we aren't outputting to the manager. An embedded CR-NL in a CLI command screws up several AMI parsers that don't expect to see that combination in the middle of output. (Closes issue #14305) Reported by: martins Patch by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	426cee0362	This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed. Fixed for snuff-home on -dev channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	a5198f55e0	Fixes issue with undefined audio codecs in chan_iax2 During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-15 are not defined, these bits are never turned off. In trunk, bits 13-15 are defined, which means 1.4 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue #14283) Reported by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

1 2 3 4 5 ...

13328 Commits (3cbc42e2e458dac6ccc68e68ada7dfcb416fd111) All Branches Search

13328 Commits (3cbc42e2e458dac6ccc68e68ada7dfcb416fd111)

All Branches