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Merged revisions 369871 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369871 | kmoore | 2012-07-10 08:35:30 -0500 (Tue, 10 Jul 2012) | 12 lines Improve Goto and GotoIf related documentation Correct documentation on labeliftrue and labeliffalse parameters of GotoIf() and update several other locations that use the same syntax. (closes issue ASTERISK-20007) Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged revisions 36986…
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10.7.0-digiumphones-rc1
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Add support for exposing the received contact URI and also for setting the request URI in messages. (closes issue AST-911)
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Merged revisions 369819 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369819 | qwell | 2012-07-09 12:06:40 -0500 (Mon, 09 Jul 2012) | 9 lines Add Digium phones context to sip_notify sample config. This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) ........ Merged revisions 369818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................
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Merged revisions 369793 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369793 | jrose | 2012-07-09 09:43:49 -0500 (Mon, 09 Jul 2012) | 9 lines chan_sip: Fix small behavioral change accidentally introduced in r369750 When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed the return value, which would likely make the indication not be sent in audio. This fixes that while still removing the warning m…
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Merged revisions 369751 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369751 | jrose | 2012-07-06 16:02:37 -0500 (Fri, 06 Jul 2012) | 12 lines chan_sip: Add case for FLASH control frames so that we don't display a warning. chan_sip channels can receive flash control frames when connected to analog phones and possibly for other reasons. There really isn't a reason to warn when these frames are received, we can safely ignore them. Patc…
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Merged revisions 369732 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369732 | mmichelson | 2012-07-06 13:47:05 -0500 (Fri, 06 Jul 2012) | 21 lines Remove a superfluous and dangerous freeing of an SSL_CTX. The problem here is that multiple server sessions share a SSL_CTX. When one session ended, the SSL_CTX would be freed and set NULL, leaving the other sessions unable to function. The code being removed is superfluous because the …
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Merged revisions 369709 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369709 | mmichelson | 2012-07-06 10:23:28 -0500 (Fri, 06 Jul 2012) | 14 lines Fix bridging thread leak. The bridge thread was exiting but was never being reaped using pthread_join(). This has been fixed now by calling pthread_join() in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012 …
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Merged revisions 369653 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369653 | kmoore | 2012-07-05 14:12:33 -0500 (Thu, 05 Jul 2012) | 20 lines Resolve heap corruption issue with voicemail The heard and deleted arrays in the voicemail state structure were not handled properly following the memory leak fix in r354890 and a fix for an invalid free in r356797. This could result in accessing and writing into freed memory. The allocation…
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Merged revisions 369627 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369627 | mjordan | 2012-07-05 12:02:53 -0500 (Thu, 05 Jul 2012) | 18 lines Do not send a BYE when a provisional response arrives during a re-INVITE Commits r369557 and r369579 were done to improve handling of re-INVITEs when the UA that was supposed to receive the re-INVITE fails to respond. A limitation of those patches occurred when a UA sent a provisional respons…
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Merged revisions 369580 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369580 | twilson | 2012-07-03 12:02:18 -0500 (Tue, 03 Jul 2012) | 11 lines More improvements to re-INVITEs timing out after a provisional response There is no need to call check_pendings() on a final response to an INVITE when destroying the scheduler entry as it will be done later during normal processing. (issue ASTERISK-19992) ........ Merged revisions 3695…
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Merged revisions 369558 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369558 | twilson | 2012-07-03 09:34:22 -0500 (Tue, 03 Jul 2012) | 14 lines Better handle re-INVITEs with provisional but no final repsonses A previous attempt at fixing this issue had negative side effects related to attended transfers which this patch should resolve. Many thanks to Steve Davies for all of the good suggestions and testing. (closes issue ASTERISK-19…
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Merged revisions 369511 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ........ r369511 | mmichelson | 2012-06-29 15:28:10 -0500 (Fri, 29 Jun 2012) | 3 lines Fix apparent copy and paste error where incorrect "glue" is used. ........
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Merged revisions 369491 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369491 | file | 2012-06-29 11:54:11 -0500 (Fri, 29 Jun 2012) | 5 lines With some configurations a transport is not actually specified so assume UDP in these cases. ........ Merged revisions 369490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................
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Merged revisions 369472 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369472 | file | 2012-06-29 10:30:47 -0500 (Fri, 29 Jun 2012) | 10 lines Make the address family filter specific to the transport. (closes issue ASTERISK-16618) Reported by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........ Merged revisions 369471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................
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Merged revisions 369437 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369437 | twilson | 2012-06-27 16:10:01 -0500 (Wed, 27 Jun 2012) | 16 lines Clean up after a reinvite that never gets a final response The basic problem is that if a re-INVITE is sent by Asterisk and it receives a provisional response, but no final response, then the dialog is never torn down. In addition to leaking memory, this also leaks file descriptors and will e…
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Merged revisions 369391 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369391 | mjordan | 2012-06-26 08:22:42 -0500 (Tue, 26 Jun 2012) | 15 lines 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 derefer…
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Merged revisions 369369 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369369 | mjordan | 2012-06-25 14:36:02 -0500 (Mon, 25 Jun 2012) | 29 lines 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…
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Merged revisions 369353 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369353 | mmichelson | 2012-06-25 14:16:52 -0500 (Mon, 25 Jun 2012) | 14 lines 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) reporte…
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Merged revisions 369325,369328 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369325 | mmichelson | 2012-06-25 10:52:42 -0500 (Mon, 25 Jun 2012) | 20 lines Multiple revisions 369323-369324 ........ r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines 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 f…
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Merged revisions 369303 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369303 | mmichelson | 2012-06-25 09:23:16 -0500 (Mon, 25 Jun 2012) | 14 lines 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. (C…
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Merged revisions 369283 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369283 | rmudgett | 2012-06-22 19:12:27 -0500 (Fri, 22 Jun 2012) | 22 lines 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. * …
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Merged revisions 369259,369263 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369259 | rmudgett | 2012-06-22 16:37:05 -0500 (Fri, 22 Jun 2012) | 5 lines Check if PBX was started and fix F and F(x) action logic in Dial application. ........ Merged revisions 369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r369263 | rmudgett | 2012-06-22 17:09:29 -0500 (Fri, 22 Jun 2012) | 5 lines Explicitly check caller hangup in…
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Merged revisions 369236,369239 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369236 | rmudgett | 2012-06-22 15:49:33 -0500 (Fri, 22 Jun 2012) | 5 lines Change incorrect chan_sip zombie hangup debug message. They are all zombies now. ........ Merged revisions 369235 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r369239 | rmudgett | 2012-06-22 16:04:25 -0500 (Fri, 22 Jun 2012) | 5 lines Check if PBX was started for …
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Merged revisions 369215 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369215 | twilson | 2012-06-22 14:34:59 -0500 (Fri, 22 Jun 2012) | 9 lines 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. ........ Merged revisions 369214 from http://svn.asterisk.org/svn/aste…
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Merged revisions 369206 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369206 | kmoore | 2012-06-22 12:23:26 -0500 (Fri, 22 Jun 2012) | 11 lines 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 vide…
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Merged revisions 369147 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369147 | may | 2012-06-20 12:36:27 -0500 (Wed, 20 Jun 2012) | 10 lines fix locking issue on empty callList (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch ........ Merged revisions 369146 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................
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Merged revisions 369109 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369109 | elguero | 2012-06-19 21:04:58 -0500 (Tue, 19 Jun 2012) | 23 lines 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 pointe…
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Merged revisions 369091 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ........ r369091 | may | 2012-06-19 18:32:06 -0500 (Tue, 19 Jun 2012) | 9 lines 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-ast10.patch ........
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Merged revisions 369067 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369067 | mmichelson | 2012-06-19 10:37:37 -0500 (Tue, 19 Jun 2012) | 17 lines 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. Ins…
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Merged revisions 369044 via svnmerge from file:///srv/subversion/repos/asterisk/branches/10 ................ r369044 | rmudgett | 2012-06-18 13:11:30 -0500 (Mon, 18 Jun 2012) | 12 lines 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 ........ Merged revisions 369043 from http://svn.asterisk.org/svn/…
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