asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	6692a0ad9e	Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel. It was strange that the AgentCalled AMI event would get most of its information from the incoming channel but then get the CallerID information from the outgoing channel. Before connected line support was added, this information was always the same at this point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham Tested by: rmudgett ........ Merged revisions 344536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	3a2651f393	Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever. When a conference user enters the DTMF menu they are suspended from the bridge while the channel is handed off to the DTMF feature code. If a user entered this state and hungup, there existed a race condition where the channel could not exit the conference because it was waiting on a signal that would never arrive. This patch fixes that, because it would stupid for me to talk about the problem and commit a patch for something else. (closes issue ASTERISK-18829) Reported by: zvision git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	bf43b7c30e	Fix another incorrect case with meetme's PIN logic and add documentation This fixes an issue where a user of a dynamic conference was asked for a PIN twice. This also adds documentation to assist in future modifications to the piece of code responsible for PIN checking. (closes issue AST-670) ........ Merged revisions 344439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	f20fd93eff	Fix several bugs with SDP parsing and well-formedness of responses Fix bug ASTERISK-16558 which dealt with the order of responses to incoming streams defined by SDP. Fix unreported bug where offering multiple same-type streams would cause Asterisk to reply with an incorrect SDP response missing one or more streams without a proper declination. Fix bugs related to a single non-audio stream being offered with responses requesting codecs that were not offered in the initial invite along with an additional audio stream that was not in the initial invite. Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged revisions 344385 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	248bf67ae5	only attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses Patch by: jkonieczny (modified) ASTERISK-18490 ........ Merged revisions 344330 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	eca7e36337	Fix deadlock during dialplan reload. Another deadlock between the conlock/hints and channels/channel locking orders. * Don't hold the channel and private lock in sip_new() when calling ast_exists_extension(). (closes issue ASTERISK-18740) Reported by: Byron Clark Patches: sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark Tested by: Byron Clark ........ Merged revisions 344268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	11a6abe970	Don't treat a host:port string as a domain The domain matching code prior to 1.8 used to manually remove the port from the host:port string when determining if an incoming request matched the list of domains. When switching to the new parsing functions, the documentation implied that the "domain" was being returned by these functions, when instead it was returning the "hostport" as defined by RFC 3261. This led to confusion and resulted in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch renames the "domain" variables in the parsing functions to "hostport" to more accurately describe what it is that they are returning and also properly truncates the resulting hostport strings when dealing with domain matching. Review: https://reviewboard.asterisk.org/r/1574/ ........ Merged revisions 344215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	4067667855	Add a unit test for ast_sockaddr_split_hostport Review: https://reviewboard.asterisk.org/r/1575/ ........ Merged revisions 344157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	c27b331d0f	delete svn:mergeinfo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	b41b71f0e5	Generate response to Status Enquiry message with Status q.931 message. Some PBXes require this for call status checking (closes issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches: ASTERISK-18748-5.patch (License #5415) patch uploaded by may213 Tested by: Fabrizio Lazzaretti ........ Merged revisions 344158 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	42686b23da	Fix pin parameter behavior regression in MeetMe The last time this code was touched (by me), a subtlety was missed based on the difference between needing to check a pin's validity and the need to prompt for a pin. (closes issue ASTERISK-18488) ........ Merged revisions 344102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	cf5149dd31	don't call ltohl() twice on the same value ASTERISK-18739 Patch by: pawel (modified) ........ Merged revisions 344048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a7a692c09f	Residual changes for Asterisk v10 branch from ASTERISK-18747. Residual changes for Asterisk v10 branch from ASTERISK-18747 after https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs callid hash key change fix. * Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from dialogs_rtpcheck. This is an optimization to avoid an unneeded lock/unlock and object search when using ao2_unlink. * Prevent crash in check_rtp_timeout() if dialog->rtp is NULL. Review: https://reviewboard.asterisk.org/r/1557/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344004 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	09e9df32eb	Fix crash when dialplan remove include is called with too few arguments. "dialplan remove include x from y" crashed when the amount of arguments was less than 6. (closes issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by: Andrey Solovyev ........ Merged revisions 343936 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	51fb836f56	Fixes regression caused by r343635 There was a missing unlock for a function return that is only present in Asterisk 10 and Asterisk Trunk. (closes issue ASTERISK-18839) Reported by: Michael L. Young Patches: asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4bd0ef0744	Fixed reference to incorrect variable if unknown host configured crash. * Fixed a LOG_ERROR message referencing the config variable list v that had previously been processed and became NULL. * Added error return value set that was missing in an ast_append_ha() error return path. (closes issue ASTERISK-18743) Reported by: Michele Patches: issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes Tested by: Michele ........ Merged revisions 343851 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7602fbc0d9	Recorded merge of revisions 343791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Fix boo-boo in prep_tarball script. A hardcoded a branch number was in the prep_tarball which could not work. Changed it to the variable. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	c95fa81971	Fix boo-boo in prep_tarball script. A hardcoded a branch number was in the prep_tarball which could not work. Changed it to the variable. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	fe7ae31263	Make "sip show settings" CLI command get RPID flags from the right global page The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command pulled the flags from the incorrect global flags page. These are now read from sip global flags page 0. (closes issue AST-711) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	afad8e69e5	respect case changes in peer names on sip reload ASTERISK-18669 ........ Merged revisions 343690 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5426a95b2c	Fix __sip_subscribe_mwi_do() incorectly changing dialogs hash key callid. Changing an object value used as a container key requires removing the object from the container and reinserting it. * Created change_callid_pvt() to call instead of build_callid_pvt(). The change_callid_pvt() will correctly change the dialog callid so the ao2 conainter can explicitly unlink it. ........ Merged revisions 343637 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	ee43e5c271	Prevent BLF subscriptions from causing deadlocks Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks. This function now requires that both the peer and associated pvt be unlocked before it is called for cases where peer and peer->mwipvt form a circular reference. (closes issue ASTERISK-18663) Review: https://reviewboard.asterisk.org/r/1563/ ........ Merged revisions 343621 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	fb2dbd51dc	Correct the default udptl port range. The udptl port range was defined as 4000-4999 in the udptl.conf.sample, as 4500-4599 if you didn't have a config and 4500-4999 if your config was broken. Default is now 4000-4999. (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher Review: https://reviewboard.asterisk.org/r/1565 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	11aeccf707	Fix deadlock if peer is destroyed while sending MWI notice. A dialog cannot be destroyed by the ao2_callback dialog_needdestroy because of a deadlock between the dialogs container lock and the RWLOCK of the events subscription list. * Create dialogs_to_destroy container to hold dialogs that will be destroyed. * Ensure that the event subscription callback will never happen with an invalid peer pointer by making the event callback removal the first thing in the peer destructor callback. NOTE: This particular deadlock will not happen with Asterisk 10, but some of the changes still apply. (closes issue ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review: https://reviewboard.asterisk.org/r/1564/ ........ Merged revisions 343577 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	b65fd0ee03	list all of the codecs associated with a particular format id for CLI command "core show codec" AST-699 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	e0c755eb89	Final fix memleaks in GkClient codes, same for Timer codes. (these memleaks stop development of gk codes, now i can continue) Fix printHandler 'Unbalanced Structure' issues with locking printHandler data for single thread. ........ Merged revisions 343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	02c8aaf3e3	Fix sqlite config driver segfault and broken queries The sqlite realtime handler assumed you had a static config configured as well. The realtime multientry handler assumed that you weren't using dynamic realtime. (closes issue ASTERISK-18354) (closes issue ASTERISK-18355) Review: https://reviewboard.asterisk.org/r/1561 ........ Merged revisions 343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7ffec7ea4d	Remove invalid flag given to iterator in func_dialgroup.c ........ Merged revisions 343336 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	3acf74f67a	Make room for the fax detect flags The original REGISTERTRYING flag, in addition to being impossible to check, also encroached on the space for the flag above it. This patch moves the flags that were below REGISTERTRYING back to where they were as though we had just removed the REGISTERTRYING option. ........ Merged revisions 343276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	ec558d7d99	Remove registertrying option in chan_sip This option is not only useless, but has been broken since inception since the flag was never copied from the peer where it is set to the pvt where it was checked. RFC 3261 specificially states that you should not send a provisional response to a non-INVITE request, and if we did fix the code so that it worked, it would cause the same kind of user enumeration vulnerability that we've discussed with the nat= setting. This patch removes registertrying option and any code that would have sent a 100 response to a register. Review: https://reviewboard.asterisk.org/r/1562/ ........ Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	83bda07fed	Fix improper warning introduced by r342927 and more tweaks Changeset r342927 introduced a warning which was only supposed to be emitted when a found realtime peer had an empty (or no) name. It turned out that there were some inconsistencies left. Now found peers with an empty name are explicitly ignored like before r342927 but better. Reviewed by: Stefan Schmidts, Terry Wilson Review: https://reviewboard.asterisk.org/r/1560 ........ Merged revisions 343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	e04eea4fab	Ensure that string field lengths are properly aligned Integers should always be aligned. For some platforms (ARM, SPARC) this is more important than for others. This changeset ensures that the string field string lengths are aligned on all platforms, not just on the SPARC for which there was a workaround. It also fixes that the length integer can be resized to 32 bits without problems if needed. (closes issue ASTERISK-17310) Reported by: radael, S Adrian Reviewed by: Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review: https://reviewboard.asterisk.org/r/1549 ........ Merged revisions 343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	3769a0e929	Add note about how Authenticate() application with option 'd' works. (closes issue ASTERISK-17422) Reported by: Leif Madsen ........ Merged revisions 343102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	3e76d988a5	Update documentation for leastrecent strategy. In queues.conf.sample the leastrecent strategy was incorrectly described. Now updated to reflect how the strategy actually checks peers. (closes issue ASTERISK-17854) Reported by: Sebastian Denz Patches: queues.conf-doc_issue.patch (License #6139) ........ Merged revisions 343047 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	48818ce81b	Modify comments in MeetMe application documentation about DAHDI. The MeetMe application documentation has some comments about usage of DAHDI, and they were a bit outdated relative to modern DAHDI releases. This patch changes the comment to just tell the user that a functional DAHDI timing source is required, and no longer mention 'dahdi_dummy', since that module does not exist in current DAHDI releases. ........ Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	14200ca67f	Several fixes to the chan_sip dynamic realtime peer/user lookup There were several problems with the dynamic realtime peer/user lookup code. The lookup logic had become rather hard to read due to lots of incremental changes to the realtime_peer function. And, during the addition of the sipregs functionality, several possibilities for memory leaks had been introduced. The insecure=port matching has always been broken for anyone using the sipregs family. And, related, the broken implementation forced those using sipregs to still have an ipaddr column on their sippeers table. Thanks Terry Wilson for comprehensive testing and finding and fixing unexpected behaviour from the multientry realtime call which caused the realtime_peer to have a completely unused code path. This changeset fixes the leaks, the lookup inconsistenties and that you won't need an ipaddr column on your sippeers table anymore (when you're using sipregs). Beware that when you're using sipregs, peers with insecure=port will now start matching! (closes issue ASTERISK-17792) (closes issue ASTERISK-18356) Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1395 ........ Merged revisions 342927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342929 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	ef82f32ca3	Cleanup references to sipusers and sipfriends dynamic realtime families Somewhere between 1.4 and 1.8 the sipusers family has become completely unused. Before that, the sipfriends family had been obsoleted in favor of separate sipusers and sippeers families. Apparently, they have been merged back again into a single family which is now called "sippeers". Reviewed by: irroot, oej, pabelanger Review: https://reviewboard.asterisk.org/r/1523 ........ Merged revisions 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	96d21568fa	Misc format capability fixes. * Fixed typo in format_cap.c:joint_copy_helper() using the wrong variable. * Fix potential race between checking if an interface exists and adding it to the container in format.c:ast_format_attr_reg_interface(). * Fixed double rwlock destroy in format.c:ast_format_attr_init() error exit path. * Simplified format.c:find_interface() and format.c:has_interface(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	7ea3892aef	Fixed invalid memory access when adding extension to pattern match tree When an extension is removed from a context, its entry in the pattern match tree is not deleted. Instead, the extension is marked as deleted. When an extension is removed and re-added, if that extension is also a prefix of another extension, several log messages would report an error and did not check whether or not the extension was deleted before accessing the memory. Additionally, if the extension was already in the tree but previously deleted, and the pattern was at the end of a match, the findonly flag was not honored and the extension would be erroneously undeleted. Additionaly, it was discovered that an IAX2 peer could be unregistered via the CLI, while at the same time it could be scheduled for unregistration by Asterisk. The unregistration method now checks to see if the peer was already unregistered before continuing with an unregistration. (closes issue ASTERISK-18135) Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1526 ........ Merged revisions 342769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	38b76521d2	Don't crash on empty notify channel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	24a8f7627d	Fix AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable. AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an iteration or before AST_LIST_REMOVE_CURRENT() without corrupting the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the list if AST_LIST_INSERT_BEFORE_CURRENT() or AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed cut and paste error using the wrong variable in AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	16a68810bf	tweak the v21 detector to detect an additional pattern of hits and misses git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	5f9c4b212d	Fix sequence number overflow over 16 bits causing codec change in RTP packets. Sequence number was handled as an unsigned integer (usually 32 bits I think, more depending on the architecture) and was put into the rtp packet which is basically just a bunch of bits using an or operation. Sequence number only has 16 bits allocated to it in an RTP packet anyway, so it would add to the next field which just happened to be the codec. This makes sure the sequence number is set to be a 16 bit integer regardless of architecture (hopefully) and also makes it so the incrementing of the sequence number does bitwise or at the peak of a 16 bit number so that the value will be set back to 0 when going beyond 65535 anyway. (closes issue ASTERISK-18291) Reported by: Will Schick Review: https://reviewboard.asterisk.org/r/1542/ ........ Merged revisions 342602 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	7106cd9f2b	Cleanup reference leaks in res_jabber res_jabber.c had a number of places where astobjs would be referenced and have their reference counts bumped without having a dereference made before the object lost scope. This patch adds a number of ASTOBJ_UNREFs to resolve that. Review: https://reviewboard.asterisk.org/r/1478/ ........ Merged revisions 342545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1a6c26b20e	Check fopen return value for ao2 reference debug output. Reported by: wdoekes Patched by: wdoekes Review: https://reviewboard.asterisk.org/r/1539/ ........ Merged revisions 342487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b58a357fde	Change D-channel warning to be less confusing on non-NFAS setups. The "No D-channels available! Using Primary channel as D-channel anyway!" WARNING message has been confusing on non-NFAS setups. The message refers to things that are NFAS specific. * Changed the warning to several different warnings to be more accurate for the situation and less confusing as a result: "No D-channels up! Switching selected D-channel from X to Y.", "No D-channels up!", and "D-channel is down!". ........ Merged revisions 342484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	ba2c590c18	Use int for storing ao2_container_count instad of size_t AST-676 ........ Merged revisions 342435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	7e46d3ee91	Simplify queue membercount code Despite an ominous sounding comment stating that membercount was for "logged in" members only and thus we couldn't use ao2_container_count(), I could not find a single place in the code where that seemed to be accurate. The only time we decremented membercount was when we were marking something dead or actually removing it. The only places we incremented it were either after ao2_link(), or trying to correct for having set it to 0 during a reload. In every case where we were correcting the value, it seemed that we were trying to make the count actually match what ao2_container_count() would return. The only place I could find where we made a determination about something being "logged in" or not, we didn't trust the membercount, but instead looked at devicestate, paused, etc. This patch removes membercount, replaces its use with ao2_container_count, and manually adds the results of ao2_container_count to a "membercount" field for ast_data queue query results. This patch also would fix AST-676, but as it is slightly riskier than the previously committed fix, the two commits have been made separately. Reivew: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	1bc124ae58	Properly update membercount for reloaded members Since q->membercount is set to 0 before reloading, it is important to increment it again for reloaded members as well as added. (closes issue AST-676) Review: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	4d81b7889d	Fix compilation on Snow Leopard/FreeBSD for pbx_spool.c One of the changes in the recent spool handling of hardlinks patch was just outside a HAVE_INOTIFY block and caused compilation to fail in some build environments. This has been corrected. ........ Merged revisions 342328 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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