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${ noResults }
22061 Commits (02fb0d8d28c123a9b998f78a77d0e33d30a80f2c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
02fb0d8d28 |
Fix crash in unloading of res_adsi module
When res_adsi is unloaded, it removes the ADSI functions that it previously installed by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs. This function was not checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing it to check whether or not the version of the functions matches what the core was expecting it. This patch makes it so that the version is only checked if a potentially valid adsi_funcs pointer was passed in. Passing in NULL removes the installed functions, bypassing the version check. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369390 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
5a77cbe00c |
Tweak CDR change in r369351
As Tilghman pointed out on review 1996, the check to see if a CDR end time has been set is sufficient to know whether or not the duration value can be used. The check-in done for r369351 forgot to include this change. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
9b9a84740b |
Re-fix how local tag is generated when sending a 481 to an INVITE.
Match our local tag to whatever to-tag was sent in the initial INVITE. Because the size of the to-tag may not fit in the buffer in the sip_pvt, it has been changed to a string field. (closes issue ASTERISK-19892) reported by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369352 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
87e212e120 |
Fix incorrect duration reporting in CDRs created in batch mode
Certain places in core/cdr.c would, if the duration value were 0, calculate the duration as being the delta between the current time and the time at which the CDR record was started. While this does not typically cause a problem in non-batch mode, this can cause an issue in batch mode where CDR records are gathered and written long after those calls have ended. In particular, this affects calls that were never answered, as those are expected to have a duration of 0. Often, this would result in CDR logs with a significant number of calls with lengthy durations, but dispositions of "BUSY". Note that this does not affect cdr_csv, as that backend does not use ast_cdr_getvar and instead directly reports the duration value. The affected core backends include cdr_apative_odbc and cdr_custom; other extended or deprecated CDR backends may potentially still directly manipulate the duration values. (issue ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883) Reported by: Thomas Arimont Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1996/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369351 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
dd1390e777 |
Fix Bridge application occasionally returning to the wrong location.
* Fix do_bridge_masquerade() getting the resume location from the zombie channel. The code must not touch a clone channel after it has masqueraded it. The clone channel has become a zombie and is starting to hangup. (closes issue ASTERISK-19985) Reported by: jamicque Patches: jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
65c1d24e29 |
Forgot to svn add this file in my last commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369324 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
bd11706d8e |
Eliminate embedding of res_adsi.so module.
The way this is done is to stop using the optional API. Instead, res_adsi.so, when loaded fills in a table of function pointers. Review: https://reviewboard.asterisk.org/r/1991 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369323 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
37a606f526 |
Be more consistent with the return code for requests received from invalid domain.
When Asterisk receives an INVITE from an external domain when allowexternaldomains=no send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving a REGISTER in this situation. (Closes issue ASTERISK-19601) Reported by Matthew Jordan Patches: ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369302 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
8d8f318272 |
Fix Bridge application and AMI Bridge action error handling.
* Fix AMI Bridge action disconnecting the AMI link on error. * Fix AMI Bridge action and Bridge application not checking if their masquerades were successful. * Fix Bridge application running the h-exten when it should not. * Made do_bridge_masquerade() return if the masquerade was successful so the Bridge application and AMI Bridge action could deal with it correctly. * Made bridge_call_thread_launch() hangup the passed in channels if the bridge_call_thread fails to start. Those channels would have been orphaned. * Made builtin_atxfer() check the success of the transfer masquerade setup. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369282 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
5e54864c5c |
Explicitly check caller hangup in app Queue rather than a polluted res2 value.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
9f3ce626fe |
Check if PBX was started and fix F and F(x) action logic in Dial application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369258 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
8139102c83 |
Check if PBX was started for generic CCSS recall.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369238 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
3fd1ede7fc |
Change incorrect chan_sip zombie hangup debug message. They are all zombies now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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4151dff715 |
Don't crash on a guest directmedia call
A sip_pvt may not have relatedpeer set if a call doesn't match up with a peer. If there is no relatedpeer, there is no direct media ACL to apply, so just return that it is allowed. (closes issue ASTERISK-20040) Reported by: Terry Wilson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369214 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
c842a8c145 |
Don't parse media stream state for SIP video streams
The sendonly/recvonly/sendrecv/inactive media stream attributes were parsed for video, but nothing was ever done with them. With this code removed, an UNSUPPORTED message is produced when these attributes are used in conjunction with a video stream which is the better behavior since they were never really supported in the first place. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369195 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
c1bafc8f9f |
fix locking issue on empty callList
(issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
71fbf1a748 |
fix compile error (1.8 don't have ast_channel_name macro)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369130 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c0e57ac429 |
Fix NULL pointer segfault in ast_sockaddr_parse()
While working with ast_parse_arg() to perform a validity check, a segfault occurred. The segfault occurred due to passing a NULL pointer to ast_sockaddr_parse() from ast_parse_arg(). According to the documentation in config.h, "result pointer to the result. NULL is valid here, and can be used to perform only the validity checks." This patch fixes the segfault by checking for a NULL pointer. This patch also adds documentation to netsock2.h about why it is necessary to check for a NULL pointer. (Closes issue ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1990/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
1ab70391f8 |
check rtptimeouts in ooh323 channels as per config file
(rtp voice, video, udptl except rtcp) (closes issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches: 19179-ooh323-2.patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
5e5564b041 |
Fix request routing issue when outboundproxy is used.
Asterisk was incorrectly setting the destination of CANCELs and ACKs for error responses to the URI of the initial INVITE. This resulted in further requests, such as INVITEs with authentication credentials, to be routed incorrectly. Instead, when these CANCEL or ACKs are to be sent, we should simply keep the destination the same as what it previously was. There is no need to alter it any. (closes issue ASTERISK-20008) Reported by Marcus Hunger Patches: ASTERISK-20008.patch uploaded by Mark Michelson (license #5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369066 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
239470bc65 |
Fix monitoring calls put in a parking lot.
* Fix a regression that was introduced by -r366167 which effectively disabled monitoring parked calls. (closes issue ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369043 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
977a9791be |
Add a script to enable finding source files without support-levels defined.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369002 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
f83d1b98e8 |
Add support-level indications to many more source files.
Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369001 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6738e63081 |
Revert Makefile change to remove embedding res_adsi.so
The change has resulted in a linking error for certain versions of GCC. This is much worse than the original issue, so for now, temporarily revert the change. A more thorough change will be sought out. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368927 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
a64586e5c2 |
Fix a deadlock that occurs when func_volume is used on a local channel.
This was discovered by trying to perform a call forward to an extension that makes use of func_volume. When the local channel is optimized away, the datastore on the local;2 channel would have its audiohook destroyed rather than detaching the audiohook from the channel and then destroying it. With this patch, func_volume's datastore destructor takes the proper route of detaching the audiohook and then destroying it. (closes issue ASTERISK-19611) reported by Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark Michelson (license #5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368898 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
89ff2d1ed7 |
Mark res_smdi/res_adsi as 'core' supported modules
Recently, various issues surrounding weak symbols have caused problems with modules that rely on that feature to be enabled in menuselect. This includes app_voicemail and chan_dahdi, as they both rely upon res_smdi and res_adsi, which, in certain circumstances, may not be enabled by default in menuselect. Because res_smdi/res_adsi are dependencies for chan_dahdi/app_voicemail, this patch marks both as 'core' supported modules. This will allow both app_voicemail and chan_dahdi to be enabled as well, regardless of whether or not that system supports weak symbols. (issue AST-900) Reported by: Thomas Arimont (issue AST-885) Reported by: Denis Alberto Martinez git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368894 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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b680746454 |
Remove forced linking of res_adsi.o
In GCC 4.5+ the result is that Asterisk has a phantom module loaded at startup, claiming to be res_adsi. (closes issue ASTERISK-19920) reported by Leif Madsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
14b242451b |
Do not install empty directories; add ASTLIBDIR
r368830 modified the installation script to only create a directory if that directory does not exist. If some directory variable was empty, it would attempt to create the empty location. It also failed to create the ASTLIBDIR directory. This patch fixes it such that the correct directories are made and only created if a value specifying them actually exists. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368852 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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1a79e238a3 |
Do not perform install on existing directories
If a directory already exists, performing a 'make install' will remove the permissions associated with the current directory and replace them with the permissions of the user executing the install. This patch changes this behavior to only perform an install on the directory if the directory does not exist. Thus, if a user later changes the permissions on that directory, those permissions will be preserved in subsequent installs. Review: https://reviewboard.asterisk.org/r/1986 Review: https://reviewboard.asterisk.org/r/1864 (closes issue ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger, Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded by mjordan) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368830 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
5919aa6cf4 |
Set the Caller ID "tag" on peers even if remote party information is present.
On incoming calls, we were setting the cid_tag on the dialog only if there was no remote party information (Remote-Party-ID or P-Asserted-Identity) present. The Caller ID tag is an invented parameter, though, and should be set no matter the circumstance. (closes issue ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884) Reported by Trey Blancher git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368807 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
3863d34cdf |
Fix deadlock potential with ast_set_hangupsource() calls.
Calling ast_set_hangupsource() with the channel lock held can result in a deadlock because the function also locks the bridged channel. (issue ASTERISK-19537) (closes issue ASTERISK-19801) Reported by: Alec Davis (closes issue AST-891) Reported by: Guenther Kelleter Tested by: Guenther Kelleter git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
0353a57671 |
Fix coverity UNUSED_VALUE findings in core support level files
Most of these were just saving returned values without using them and in some cases the variable being saved to could be removed as well. (issue ASTERISK-19672) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368738 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
44e963a550 |
Fix compilation in dev-mode
Backport a compilation fix in md5.c from trunk that only showed up in dev-mode under certain compiler versions. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368719 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
aa8ed90f30 |
Fix POTS flash hook to orignate a second call deadlock.
A deadlock can occur when a POTS phone tries to flash hook to originate a second call for 3-way or transfer. If another process is scanning the channels container when the POTS line flash hooks then a deadlock will occur. * Release the channel and private locks when creating a new channel as a result of a flash hook. (closes issue ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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e52f2967de |
Fix a specific scenario where ACKs are not matched.
If a dialog-starting INVITE contains a to-tag, then Asterisk will respond with a 481. In this case, the resulting incoming ACK would not be matched, so Asterisk would continue retransmitting the 481 until the transaction times out. There were two issues. Asterisk, upon creating a sip_pvt would generate a local tag. However, when the time came to transmit the 481, since there was a to-tag in the INVITE, Asterisk would place this original to-tag in the 481 response. When the ACK came in, Asterisk would attempt to match the to-tag in the ACK to the generated local tag. Unfortunately, Asterisk never actually transmitted a response with the generated local tag, so the to-tag in the ACK would not match. The other problem was that when the 481 was sent, nothing was set on the sip_pvt to indicate what CSeq is expected in the ACK. To fix the first problem, we zero out the to-tag seen in the incoming INVITE. This way, Asterisk, when time to send a response, will send its generated local tag instead. To fix the second problem, we set the sip_pvt's pendinginvite to the CSeq of the INVITE when we send a 481. (closes issue ASTERISK-19892) Reported by Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368625 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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8a0f2ec66a |
Add feature modifier to versions produced from branches
Certain branches, such as Certified Asterisk, may have a modifier added to them that specifies the features available in that branch. For branches, this modifier is expected to be reflected in the location of the branch in subversion. For example, a subversion of URL of /certified/branches/1.8.11 would have a feature modifier of 'certified'. This is slightly different then how features are determined for tags, where the feature is part of the actual tag name, e.g., "10.5.0-digiumphones". In keeping with the nomenclature used for tags, the feature specifier for branches is translated and placed after the revision numbers. For the example given previously, this would result in a branch version of "Asterisk SVN-branch-1.8.11-cert-rXXXXXX". git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368604 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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b563128877 |
Ensure overlapping hold flags do not conflict
When changing between different modes of hold, the flags were not being cleared out properly causing a failure to change hold states. (closes issue ASTERISK-19919) Patch-by: Morten Tryfoss Reported-by: Morten Tryfoss git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368586 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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eae448bc55 |
Fix parked call performing a DTMF blind transfer after being retrieved.
When a parked call was retrieved from the parking lot, it could not do a blind transfer because it caused the involved calls to be hung up unconditionally. * Made the ParkedCall application return the ast_bridge_call() return value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368567 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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ca256c003b |
Resolve some build warnings
My newly upgraded compiler caught these usages of uninitialized values. They weren't actually used. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6c63f45326 |
Ensure that pages and emails are sent using RFC822-compliant date format
When localization was added to app_voicemail, these headers were altered when they should have remained in en_US format for RFC compliance. This reverts the changes to those two lines. (closes issue ASTERISK-19876) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368520 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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503fc9458a |
Relay proper SIP responses on calling side.
Revision 351130 broke corect HANGUPCAUSE setting for the 404 case in chan_sip. Other cases were also potentially broken. This patch fixes the relaying of causes to be what they used to be. (closes issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter Doekes (via a reviewboard test to be committed later) Patches: chan_sip.diff uploaded by Pavel Troller (license #6302) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c44df31f10 |
Document BLINDTRANSFER behavior change.
(issue ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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0db9327800 |
Fix potential deadlock between masquerade and chan_local.
* Restructure ast_do_masquerade() to not hold channel locks while it calls ast_indicate(). * Simplify many calls to ast_do_masquerade() since it will never return a failure now. If it does fail internally because a channel driver callback operation failed, the only thing ast_do_masquerade() can do is generate a warning message about strange things may happen and press on. * Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This change fixes half of the deadlock reported in ASTERISK-19801 between masquerades and chan_iax. (closes issue ASTERISK-19537) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1915/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368405 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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81e1b36378 |
Fix deadlock when Gosub used with alternate dialplan switches.
Attempting to remove a channel from autoservice with the channel lock held will result in deadlock. * Restructured gosub_exec() to not call ast_parseable_goto() and ast_exists_extension() with the channel lock held. (closes issue ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368308 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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871cef109a |
Improve SDP parsing warning messages
* 'Unsupported media type' is only reported when that is in fact the case, not when a supported media type is included in an 'm' line that has an invalid format. * All warning messages related to parsing 'm' lines now include the 'm' line contents. * (minor bugfix) newline added to port-number-zero warning messages. * Warning messages improved to use RFC-specified terminology for various items. * Warnings for offers that include more than one port for a single media type now include the media type. Review: https://reviewboard.asterisk.org/r/1811/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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751a8bb6b3 |
Add documentation to function CHANNEL for options echocan_mode and buffers
The ability to set "echocan_mode" and "buffers" through the dialplan was added to chan_dahdi some time ago. This patch adds some documentation to func_channel. (Closes issue ASTERISK-19911) Reported by: Dale Noll Tested by: Michael L. Young Patches: asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1949/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
bd85d458a2 |
Coverity Report: Fix issues for error type REVERSE_INULL (core modules)
* Fixes findings: 0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
f7d2d4601b |
Use the DEADLOCK_AVOIDANCE() macro instead.
(issue ASTERISK-19854) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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0bad4cbdde |
Fix deadlock when executing CLI "pri show channels" and "ss7 show channels" commands.
* Fix sig_pri_lock_owner() to avoid deadlock properly. * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid deadlock properly. * Code ss7_grab() better. (closes issue ASTERISK-19854) Reported by: Jaxon Patches: jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded by rmudgett (Modified to do the same thing to sig_ss7) Tested by: Jaxon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367976 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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ba26e7e249 |
Coverity Report: Fix issues for error type REVERSE_INULL (deprecated modules)
* Fix only issue pointed out by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user(). * Change use of %i to %d in sscanf() in find_user(). The use of %i gives unexpected parsing because it can accept hex, octal, and decimal integer formats. * Changed other uses of %i in app_meetme() to use %d for consistency. (issue ASTERISK-19648) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |