18174 Commits (f1dc620467d9b218ad5041b6416bb030aec9c68c)

 

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Author	SHA1	Message	Date
Sean Bright	7ee6e9f4ce	Add a missing unref for queues in handle_statechange. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	3abe8a963e	Add new ast_complete_applications function so that we can use it with the ... 'channel originate ... application <app>' CLI command. (And yeah, I cleaned up some whitespace in res_clioriginate.c... big whoop, wanna fight about it!?) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	9a83ed9d93	Use a properly allocated channel for substitution in cdr_sqlite3_custom. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196725 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	d4efe15c09	Fix a bug where the sip unregister CLI command did not completely unregister the peer. ... (closes issue #15118) Reported by: alecdavis Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	02401da266	Merged revisions 196657 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7 lines Remove some bash specific stuff from safe_asterisk. (closes issue #10812) Reported by: paravoid Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license 46) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b543887664	Use a properly allocated channel for substitution in cdr_manager. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	5518c1b171	Move AGI static documentation to the new AstXML form. ... Move AGI commands documentation to XML docs: 'set priority' 'set variable' 'stream file' 'control stream file' 'tdd mode' 'verbose' 'wait for digit' 'speech create' 'speech set' 'speech destroy' 'speech load grammar' 'speech unload grammar' 'speech activate grammar' 'speech deactivate grammar' 'speech recognize' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	be4798f0b3	Move static AGI commands documentation to XML. ... Move AGI commands ('say datetime', 'send image', 'send text', 'set autohangup', 'set callerid', 'set context', 'set extension') documentation to the AstXML form. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	1d28f5acd4	Fix errors in cdr_custom that cause reference errors when non-CDR variable ... substitution is done. cdr_custom was creating a ast_channel struct directly and passing it into the core for variable substition. This was fine as long as the format string contained only calls to the CDR() function. Doing something like ${EPOCH} on the other hand tried to lock the channel, which would fail and throw an error because the passed channel hadn't been allocated as an ao2 object. So now we create the dummy channel with ast_channel_alloc, and everything works as expected. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	57eedf97d0	Correct example for CLI autocompletion (generation) ... Reported by Atis on #asterisk-dev git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Moises Silva	b93c1a2df5	set MFCR2_CATEGORY just when starting the pbx ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	3982b8c71b	Call ast_stun_init() when we're initializing to get the 'stun debug set' ... commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f50bb3bfa4	SIP set outbound transport type from Registration ... In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	de6498b2d3	Don't crash if an RTP instance can't be created. This could occur when an ... invalid bindaddr was specified in gtalk.conf. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	d6759e0117	Unregister every registered application by MiniVM. ... The MinivmMWI application was not being unregistered on unload and we were not able to load again the module or reload it. (closes issue #15174) Reported by: junky Patches: unregister_minivm_mwi.diff uploaded by junky (license 177) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	ad08eeaabf	Moved static documentation to the AstXML form. ... Moved AGI commands static documentation to XML docs ('say alpha', 'say digits', 'say number', 'say phonetic', 'say date' and 'say time'). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	2c882626a0	Implement a new element in AstXML for AMI actions documentation. ... A new xml element was created to manage the AMI actions documentation, using AstXML. To register a manager action using XML documentation it is now possible using ast_manager_register_xml(). The CLI command 'manager show command' can be used to show the parsed documentation. Example manager xml documentation: <manager name="ami action name" language="en_US"> <synopsis> AMI action synopsis. </synopsis> <syntax> <xi:include xpointer="xpointer(...)" /> <-- for ActionID <parameter name="header1" required="true"> <para>Description</para> </parameter> ... </syntax> <description> <para>AMI action description</para> </description> <see-also> ... </see-also> </manager> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3c7db3b7e8	Two more minor fixes due to constification ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	fb39d11e6f	Fix res_agi compilation after the const-ify the world merge. ... Since we are dealing with a 'const char * const' now, we have to create a temporary copy of the string to work on rather than the original. Fix inspired by reporter. Reviewed by everyone-and-their-mother in #asterisk-dev. (closes issue #15184) Reported by: andrew git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ee4f11cd24	s/it's/its/ ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	76e9c034be	resolve compiler warning ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	fcda626f3c	Fix build under dev mode and remove some casts that are no longer necessary as ... a result of the const-ify the world patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	0510be3564	Fix constify the world compile problem. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	63e4b99e79	Make chan_misdn compile. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	678045fb43	Merged revisions 196116 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 lines Fix a bug where using immediate with mISDN caused a cause code of 16 to get sent back instead of 1 if the 's' extension did not exist. (closes issue #12286) Reported by: lmamane ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	bb838bc67a	Avoid using prototypes when not necessary (it is already defined in the header ... file). Make log_match_char_tree() static to main/pbx.c (only used there). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) ... This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	88bda581ec	Merged revisions 195991 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer. There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement. (closes issue #15032) Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380) Tested by: guillecabeza (closes issue #14216) Reported by: Andrey Sofronov ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	1e3ac401f4	Pass connected line updates along during a bridge. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	df4dce6837	Rework the cdr_custom.conf.sample header a bit to reflect the changes in ... functionality (allowing multiple mappings). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	d02ad6b5f7	Merged revisions 195881 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	873a46ded5	If a variable had a blank value upon the initial setting, then it would do nothing. ... Identified by Dmitry Andrianov via private email, fixed by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	56903a7485	Get rid of some duplicated code and correct a connected line error. ... When receiving a 200 OK response to an INVITE, it was possible to transmit two connected line updates instead of a single one. Furthermore, the second did not have the proper information present. Now the two have been combined into a single update and the correct information is presented. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8139c8ed4e	Plug a memory leak in app_dial. ... Since we may have copied connected line info into the chanlist struct prior to placing an outbound call, we need to be sure to free the allocated data when we hang the call up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	b4c24d2da1	Merged revisions 195688 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 lines Fix some code that wrongly assumed a pointer would always be non-NULL when dealing with CDRs after a bridge. (closes issue #15079) Reported by: barryf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4da3a150f3	Merged revisions 195635 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 lines Fix a bug where the MeetMe option 'D' did not actually prompt for the pin. (closes issue #15050) Reported by: pmhaddad ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7b4eeed257	Add basic support for handling connected line-related UPDATE requests. ... SIP purists may want to look the other way... When COLP/CONP support for SIP was committed, there was a condition under which Asterisk may transmit a SIP UPDATE in order to communicate the change in connected line information. The issue here is that while we could send a SIP UPDATE message, we were not prepared to receive such an UPDATE and would always responde with a 501 when we received an UPDATE. The situation was a bit rough. We really want to be able to receive UPDATEs having to do with connected line changes, but the amount of effort involved in properly supporting RFC 3311 was staggering. This commit represents a compromise. First, it was decided that it is important to only send a SIP UPDATE to an endpoint that is able to handle one. So, now we have added parsing of the Allow header into SIP. We store the allowed methods on SIP peers so that when we communicate with them, we already will know what we can and cannot send to them. We will parse the peer's allowed methods when he registers with us. If the peer is not the type to register with us, but the qualify option is enabled, then we will use the response to the OPTIONS request we send the peer to determine the peer's allowed methods. When the peer's registration expires, or when qualify deems the peer to be unreachable, we clear the allowed methods from the peer. For an actual call, we will copy the peer's allowed methods to the sip_pvt representing the call leg. If we are communicating with an endpoint which is not a peer, then we will just parse the Allow header from the first message we receive during the call and store the information in the sip_pvt. If, during communication with a peer, we receive a 501 response, then we will make sure to save the fact that we cannot use that method when communicating with that peer. Now, with all that infrastructure in place, the only actual place we use this information currently is when attempting to send a connected line change using an UPDATE request. If we cannot send the change immediately using an UPDATE, we will set the SIP_NEEDREINVITE flag so that we can send a REINVITE as soon as it is allowed. The second part of the changes here is for Asterisk to accept UPDATE requests that have connected line changes. Since we are not fully supporting RFC 3311, Asterisk will NOT place the UPDATE method in Allow headers it sends. Instead, if you are communicating with what you know to be another Asterisk box, you may set the rpid_update parameter in sip.conf so that we will send UPDATEs to that Asterisk box. When we send a connected line update, we set a custom header called "X-Asterisk-rpid-update." On the receiving end, if Asterisk receives an UPDATE that does not have the "X-Asterisk-rpid-update" header present, then Asterisk will respond with a 501 since media-changing UPDATEs are not supported. We should never get such UPDATEs, since as was stated earlier, Asterisk does not put UPDATE in its Allow header. If the custom header is present in the received UPDATE, though, then we will check the incoming request for connected line updates and queue the update on the channel where the change occurred. ABE-1840 ABE-1822 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c1cb7b078e	Merged revisions 195520 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009) | 7 lines Ensure thread keys are initialized before attempting to access them. (closes issue #14889) Reported by: jaroth Patches: app_voicemail.c.patch uploaded by msirota (license 758) Tested by: msirota, BlargMaN ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	99a1e0ce01	Merged revisions 195448 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 lines Fix a bug where direct RTP setup would partially occur even when disabled if the calling channel was answered. (issue #13545) Reported by: davidw (issue #14244) Reported by: mbnwa ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	bdcafc1ab4	Recorded merge of revisions 195366 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines Add a similar dependency on SMDI for voicemail as already exists for ADSI. (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	e45d681d15	Fix the CLI command 'manager show command' documentation and functionality. ... The CLI command 'manager show command' supports passing multiple action names in the same line, but it was not allowing that because of a incorrect check in the argumentes counter. Also the documentation was updated to show that this usage of the command is possible. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	9e5da09a6b	Rollback commit 195367. ... The CLI command 'manager show command' supports passing multiple AMI actions at a time. The issue with this command was in another place. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	311c997896	Avoid autocompleting passed the action name argument in the CLI command. ... When running the autocomplete of the CLI command 'manager show command <action>' it was autocompleting everything else after the <action> argument, giving an error, because this command doesn't support multiple AMI action names at a time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	75cd3f4918	Move AGI documentation from static to the XML form. ... Move the AGI commands 'receive text', 'receive char' and 'record' static documentation to XML docs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	92ace48782	Move the spawn of astcanary down, until after the call to daemon(3). ... This avoids possible conflicts with the internal implementation of daemon(3). (closes issue #15093) Reported by: tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	1413e48fca	Fix externalivr's setvariable command so that it properly sets multiple variables. ... The command had a for loop that was guaranteed to only execute once since the continuation operation of the loop would set the input buffer NULL. I rewrote the loop so that its operation was more obvious, and it would set multiple variables correctly. I also reduced stack space required for the function, constified the input string, and modified the function so that it would not modify the input string while I was at it. (closes issue #15114) Reported by: chris-mac Patches: 15114.patch uploaded by mmichelson (license 60) Tested by: chris-mac git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	c890725398	Remove some unused code. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	0163194d93	The facilityenable parameter does not have anything to do with pritimer parameters. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	dd0cd47630	Const-ify a string, fix a log message, and use the correct signature for the ... load_module function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	80bdf0f520	Merged revisions 195206 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7 lines Fix a typo which caused loss of audio when using G729 in some scenarios with a smoother present. (closes issue #15105) Reported by: bamby Patches: process-vad-correctly.diff uploaded by bamby (license 430) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago