13295 Commits (b705454875aed1fbf2e550d95dbeaed35468ea8b)

 

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Author	SHA1	Message	Date
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
Steve Murphy	d290eed8e0	This patch fixes a problem with 8-bit input to the ast_expr2 scanner. ... The real culprit was the --full argument to flex in the Makefile! This causes a 7-bit scanner to be generated. I reviewed the rules and found one rule where I needed to specifically include 8-bit chars for a token. I tested against the text supplied by ibercom, and all looks very well. This has been there a surprisingly long time! (closes issue #14498) Reported by: ibercom Patches: 14498.patch uploaded by murf (license 17) Tested by: murf git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177540 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	4675032a30	Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads. ... (closes issue #14253) Reported by: Skavin Patches: 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14) Tested by: Skavin git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	25bb888046	Force a MWI notification after subscribe request. Reported by the Resiprocate dev team. Thanks! ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	0d96c97ced	If we are able to create a speech structure unset the ERROR variable in case it was previously set. ... (issue #LUMENVOX-13) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
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