asterisk

Commit Graph

Author	SHA1	Message	Date
Steve Murphy	39a6b55183	This patch fixes a regression of sorts that was introduced in rev 24425. It basically fixes AST-190/ABE-1782. What was wrong: the user has 6000 extensions in one context; and then 6000 contexts, one per extension. The parser could only handle about 4893 of the 6000 extens in the single context. This was due to the regression I mentioned. To get rid of shift/reduce conflicts, Luigi set up right-recursive lists for globals, context elements, switch lists, and statements. Right recursive lists got rid of the warnings, but instead, they use up a tremendous amount of stack space when the lists are long. I saw this a few years back, and resolved not to fix it until someone complained. That day has arrived! After the changes were made, I ran the regression test suite, and there were no problems. I took the test case the user provided, and added 100,000 extensions to the single context, that already had 6,000 extens in it. (I'll see your 6, and raise you 100!) It takes a few minutes to read it all in, check it and generate code for it, but no problems. So, I think I can say that fundamentally, there are no longer any limits on the number of items you can place in contexts, statement blocks, switches, or globals, beyond your virt mem constraints. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	1183bf1ad0	Modify h323 to build against PTLib as well as the older PWLib Several changes in PTLib have occurred requiring build time detection. Changes accounted for include the library name change, config option change, install location change, and a boolean type change which is handled by ast_ptlib.h. Also, the sed check has been modified to properly work with autoconf >= 2.62. (closes issue #14224) Reported by: bergolth Patches: asterisk-autoconf-sed.patch uploaded by bergolth (license 661) asterisk-pwlib-v3.patch uploaded by bergolth (license 661) Tested by: jpeeler git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	c66921ec43	Document the return value of the update method (as requested on -dev list) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Doug Bailey	a591301b89	Merged revisions 177035 manually from https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 \| dbailey \| 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) \| 2 lines Fixed error where a check for an zero length, terminated string was needed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Doug Bailey	bf0b8526b1	Need to take into account the \0 terminator of the old string to determine the amount available. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Shaun Ruffell	e731e26bee	Several changes to codec_dahdi to play nice with G723. This commit brings in the changes that were living out on the svn/asterisk/team/sruffell/asterisk-1.4-transcoder branch. codec_dahdi.c now always uses signed linear as the simple codec so that a soft g729 codec will not end up being preferred to the hardware codec. There are also changes to allow codec_dahdi.c to feed packets to the hardware in the native sample size of the codec. This solves problems with choppy audio when using G723. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	de8f6bab86	Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	7799945eb9	Backport change to 1.4: Prior to masquerade, move the group definitions to the channel performing the masq, so that the group count lingers past the bridge. (closes issue #14275) Reported by: kowalma Patches: 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	e891f2a92d	After a 'sip reload', qualifies for realtime peers weren't immediately restarted, instead waiting until the next registration. We're now caching the qualify across a reload/restart and starting the qualify immediately upon loading the peer. (closes issue #14196) Reported by: pdf Patches: 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) Tested by: pdf git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	03dd54be23	Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that. issue #13749 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	4d808232cf	correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	b30800adfb	Remove unused variable and make dev-mode compilation happy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176252 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	5ae664d5aa	Open the DAHDI pseudo device and set it to be nonblocking atomically Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately from opening the file was causing an "inappropriate ioctl for device" error. While I cannot fathom why this would be happening, I certainly am not opposed to making the code a bit more compact/efficient if it also fixes a bug. (closes issue #14482) Reported by: ys Patches: meetme.patch uploaded by ys (license 281) Tested by: ys git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	1a00cbbf1d	Fixes issue with AST_CONTROL_SRCUPDATE breaking out of native bridge In iax2, when a AST_CONTROL_SRCUPDATE is received it breaks from the native bridge, but since there is no code path to handle srcupdate it just goes to be beginning of the loop. This was causing packet storms of srcupdate frames between servers. Now srcupdate frames do not break the native bridge for processing. (closes issue #13749) Reported by: adiemus git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	1b5b3efcae	fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	22734e39dc	Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	db4dc67740	fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	4c3b9ccf3b	format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	beaf6760c4	Disable format_ilbc.so by default, like codec_ilbc.so git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	ada21a8039	Make sure that the debug line is not printed on debug level 0 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	3cc3863d28	Zaptel is not DAHDI. Rather, Zaptel is actually Zaptel. (in case you're confused, DAHDI is still DAHDI) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	3e0b36d9d0	Fix a potential crash situation when using IMAP voicemail If calling into VoiceMailMain when using IMAP storage, it was possible to crash Asterisk by hanging up the phone when prompted for a voicemail mailbox. This patch fixes the issue. While it may appear that this patch is superficial, it allows code execution to continue to the failure case just below the IMAP_STORAGE code block where this patch has been applied (closes issue #14473) Reported by: dwpaul Patches: voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7accd1ec46	Fix a place where filestreams were not refcounted properly This section was already present in trunk and other branches, but did not exist in 1.4. (closes issue #14395) Reported by: ZX81 Patches: 14395.patch uploaded by putnopvut (license 60) Tested by: ZX81 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	a13deff994	Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	46963bc8b5	Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	3e396d458a	Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
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
Philippe Sultan	d045d36561	Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175029 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
Joshua Colp	9fa3324845	Go off hold when we get an empty reinvite telling us to. (closes issue #14448) Reported by: frawd Patches: hold_invite_nosdp.patch uploaded by frawd (license 610) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174644 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
Steve Murphy	680cc35607	This patch solves some compiler complaints in both 32 and 64-bit environments. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7f20e5ffab	Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e2fd8852db	Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off. (closes issue #14407) Reported by: mostyn git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174218 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c347a43c9f	Fix a race condition that could cause a crash. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Dwayne M. Hubbard	d29a99cb89	check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	c80b2b93b5	Remove a debug message I put in by accident. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	6cda579f17	Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Matthew Nicholson	5edf9d8a59	Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	76af76c5f4	Backport OS X fix from trunk (AGAIN, closes issue #14360) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	f70845aa24	Fix logic regarding when to perform an SRV lookup for outgoing REGISTER requests With this fix, we only will perform an SRV lookup at the following times: * The first time we register with a remote registrar * If we send a REGISTER but do not receive a response * If the sendto() function returns an error While I wrote the patch that fixes this issue, a huge amount of credit is due to Brett Bryant, who wrote the initial patch on which I based this one. (closes issue #12312) Reported by: jrast Patches: 12312.patch uploaded by putnopvut (license 60) Tested by: blitzrage Review: http://reviewboard.digium.com/r/132/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	0b026ce04f	Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	e7478c7b15	Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	d914df762a	Add some missing cleanup to app_mixmonitor git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	8e764c583d	Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	82937be553	Revert my previous change because it was stupid git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	fed7d2308b	Add a missing unlock. Extremely unlikely to ever matter, but it's needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173392 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

1 2 3 4 5 ...

13274 Commits (39a6b55183bb26352dca2bea3cb259eb2196e0af) All Branches Search

13274 Commits (39a6b55183bb26352dca2bea3cb259eb2196e0af)

All Branches