asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	bbc56596fd	res_pjsip_phoneprov_provider: fix compile breakage on AST_VECTOR endpoint->inbound_auths was changed to a vector in 13 and I committed the 12 patch instead of the 13 patch. Tested-by: George Joseph git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	e2ae7bd79f	res_rtp_asterisk: Crash if no candidates received for component When starting ice if there is not at least one remote ice candidate with an RTP component asterisk will crash. This is due to an assertion in pjnath as it expects at least one candidate with an RTP component. Added a check to make sure at least one candidate contains an RTP component and at least one candidate has an RTCP component. ASTERISK-24383 #close Review: https://reviewboard.asterisk.org/r/4039/ ........ Merged revisions 425030 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d3342fb98d	res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprov This module allows res_pjsip to integrate with res_phoneprov. It handles the pjsip 'phoneprov' object type. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3976/ ........ Merged revisions 425007 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cc07f4835c	res/res_phoneprov: Don't cancel Asterisk load on module load failure ........ Merged revisions 424985 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d1c9621852	res_phoneprov: Refactor phoneprov to allow pluggable config providers This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	376ec31b17	cdr.c: Make turning on CDR debug a one step process instead of two. Now "cdr set debug on" doesn't also require "core set verbose 1" to see CDR debug output. ........ Merged revisions 424941 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	4c7887b852	safe_asterisk: Don't automatically exceed MAXFILES value of 2^20. On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can exceed the per-process file limit of 2^20. This patch ensures the value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626) ........ Merged revisions 424875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424878 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424879 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0292839ae0	res_rtp_asterisk: Allow only UDP ICE candidates. The underlying library, pjnath, that res_rtp_asterisk uses for ICE support does not have support for ICE-TCP. As candidates are passed through directly to it this can cause error messages to occur when it receives something unexpected (such as a TCP candidate). This change merely ignores all non-UDP candidates so they never reach pjnath. ASTERISK-24326 #close Reported by: Joshua Colp ........ Merged revisions 424852 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424853 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	b32d8b5317	Stasis: Relegate log message to dev-mode This error message primarily applies to development tasks and will now only show up when dev-mode is enabled via configure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	9e08180304	Indexer: Format message types may not exist In Asterisk 13+, any given message type is not guaranteed to exist even if Asterisk comes up correctly since creation of the message type could be declined. The indexer should not prevent Asterisk from starting under these conditions. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	57a5e2ebee	Stasis: Only log errors for non-declined types When message type creation is declined via stasis.conf, certain operations log errors assuming that the declined type is being used before initialization or after destruction. These error messages get quite spammy for oft used message types and should not be logged in the first place since the message type is validly NULL. Reported by: Matt DiMeo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	673c2febba	data: Properly access formats in capabilities structure when adding codecs. Formats within a capabilities structure are addressed starting at 0, not 1. Assuming 1 causes it to exceed an array. ASTERISK-24389 #close Reported by: Kevin Harwell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	17d079c371	res/res_pjsip_outbound_registration: Initialize auth_reject_permanent parameter Prior to this patch, the auth_reject_permanent parameter was not initialized on the registration client state, leading to the parameter being disabled regardless of the value specified in pjsip.conf. This patch initialized the setting on the registration client state to the provided configuration value. ASTERISK-24398 #close ........ Merged revisions 424730 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9f5c73586c	res/res_pjsip_pubsub: Fix typo in WARNING message git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5c607f9735	message: Don't close an AMI connection on SendMessage action error If SendMessage encounters an error (such as incorrect input provided to the action), it will currently return -1. Actions should only return -1 if the connection to the AMI client should be closed. In this case, SendMessage causing the client to disconnect is inappropriate. This patch causes the action to return 0, which simply causes the action to fail. Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354 #close Reported by: Peter Katzmann patches: sendMessage.patch uploaded by Peter Katzmann (License 5968) ........ Merged revisions 424690 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424691 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6a8cb946eb	features.c: Fix lingering channel ref while Bridge() application is active. Using the Bridge application to bridge a channel that is executing an applicaiton such as Wait results in a lingering Surrogate channel in the CLI "core show channels" output even though it has already hungup. * Fix bridge_exec() to not hold onto the current_dest_chan ref once it has been put into the bridge. * Eliminated bridge_exec()'s use of RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged revisions 424668 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424669 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c092e49344	sdp_srtp: Add new lines to some WARNING messages ........ Merged revisions 424646 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4f28ae4f51	res_pjsip/pjsip_options: Do not 404 an OPTIONS request not sent to an endpoint An OPTIONS request that is sent to Asterisk but not to a specific endpoint is currently sent a 404 in response. This is because, not surprisingly, an empty extension is never going to be found in the dialplan. This patch makes it so that we only attempt to look up the endpoint in the dialplan if it is specified in the OPTIONS request URI. #SIPit31 ASTERISK-24370 #close Reported by: Matt Jordan ........ Merged revisions 424624 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	57233a97e8	pjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health. It will treat the channels as a PJSIP channel, eventually hitting an ao2 error, FRACKing on assertion error, and quite likely crashing. This patch adds checks to the read/write callbacks that ensure that the channel technology is of type 'PJSIP' before attempting to operate on the channel. #SIPit31 ASTERISK-24382 #close Reported by: Matt Jordan ........ Merged revisions 424621 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	69032d62aa	res_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible (although it shouldn't occur) for pjproject to pass up an rdata object with a NULL msg in the msg_info. Needless to say, things that attempt to dereference this are in for a rough ride. In particular, this caused crashes in three different locations, all of which are 'low level' enough to intercept an rdata object early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3) res_pjsip/distributor Anything that can intercept an rdata object before res_pjsip/distributor should be defensive when looking at the received packet. #SIPit31 ASTERISK-24369 #close Reported by: Matt Jordan ........ Merged revisions 424618 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f36b64f58e	res/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions A subscription that has been persisted can - for various reasons - fail to be re-created on startup. This patch resolves a number of crashes that occurred when a subscription cannot be re-created on several off-nominal paths. #SIPit31 ASTERISK-24368 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	9e3b5be182	Release AMI connections on shutdown. ASTERISK-24378 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4037/ ........ Merged revisions 424578 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424579 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	0904b18fcc	Blocked revisions 424575 ........ chan_sip: Clean leak on error path of process_sdp Resolve leak in process_sdp that occurs in 2 error path's where crypto lines are expected but not provided. ASTERISK-24385 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4045/ ........ Merged revisions 424569 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	a03464bea2	chan_motif: Correct last commit to use ao2_cleanup to free format cap This fix applies to 13 and trunk. ASTERISK-24384 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4043/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	3987b978d6	chan_motif: Release format capabilities and config on module load error ASTERISK-24384 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4043/ ........ Merged revisions 424550 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424551 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	30e6eed19d	res_pjsip: Fix XML typo and update CHANGES. ASTERISK-24199 ........ Merged revisions 424528 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	cff192429b	audiohooks: Reevaluate the bridge technology when an audiohook is added or removed. Adding a mixmonitor to a channel causes the bridge to change technologies from native to simple_bridge so the call can be recorded. However, when the mixmonitor is stopped the bridge does not switch back to the native technology. * Added unbridge requests to reevaluate the bridge when a channel audiohook is removed. * Moved the unbridge request into ast_audiohook_attach() ensure that the bridge reevaluates whenever an audiohook is attached. This simplified the mixmonitor and chan_spy start code as well. * Added defensive code to stop_mixmonitor_full() in case additional arguments are ever added to the StopMixMonitor application. * Made ast_framehook_detach() not do an unbridge request if the framehook does not exist. * Made ast_framehook_list_fixup() do an unbridge request if there are any framehooks. Also simplified the loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4046/ ........ Merged revisions 424506 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6a844be566	chan_pjsip: Fix deadlock when masquerading PJSIP channels. Performing a directed call pickup resulted in a deadlock when PJSIP channels were involved. A masquerade needs to hold onto the channel locks while it swaps channel information between the two channels involved in the masquerade. With PJSIP channels, the fixup routine needed to push a fixup task onto the PJSIP channel's serializer. Unfortunately, if the serializer was also processing a task that needed to lock the channel, you get deadlock. * Added a new control frame that is used to notify the channels that a masquerade is about to start and when it has completed. * Added the ability to query taskprocessors if the current thread is the taskprocessor thread. * Added the ability to suspend/unsuspend the PJSIP serializer thread so a masquerade could fixup the PJSIP channel without using the serializer. ASTERISK-24356 #close Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/4034/ ........ Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	b67094624d	sorcery: Prevent SEGV in sorcery_wizard_create when there's no create function When you call ast_sorcery_create() you don't necessarily know which wizard is going to be invoked. If it happens to be a wizard like 'config' that doesn't have a 'create' virtual function you get a segfault in the sorcery_wizard_create callback. This patch catches the null function pointer, does an ast_assert, and logs an error. Review: https://reviewboard.asterisk.org/r/4044/ ........ Merged revisions 424447 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424448 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	ef2c567597	PJSIP: Restore functional default for callerid_privacy The pjsip config option default fixups from r424263 altered the functional default from "allowed_not_screened" to "allowed". This change restores the functional default value when none is provided. ........ Merged revisions 424426 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	1cb36afce3	Manager: Add missing fields and documentation for CoreShowChannels This corrects some issues introduced in the responses to the CoreShowChannels AMI command as well as adding documentation for the responses. The command in Asterisk 12 was missing the following fields: Duration, Application, ApplicationData, and BridgedChannel and BridgedUniqueID (replaced with BridgeId). ASTERISK-24262 #close Reported by: Mitch Claborn Review: https://reviewboard.asterisk.org/r/4040/ ........ Merged revisions 424423 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	6246189df7	res_pjsip_session: Reduce SDP size by removing duplicate connection lines. Due to the architecture of how media streams are handled each individual handler adds connection details (IP address) for it. The first media stream is then used as the top level SDP connection line. In practice each line ends up being the same so to reduce the SDP size stream-level connection information is also added to the SDP if it differs from the top level SDP connection line. ........ Merged revisions 424414 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	94105b30a6	res_pjsip: Make transport cipher option accept a comma separated list of cipher names. Improvements to the res_pjsip transport cipher option. * Made the cipher option accept a comma separated list of OpenSSL cipher names. Users of realtime will be glad if they have more than one name to list. * Added the CLI command 'pjsip list ciphers' so a user can know what OpenSSL names are available for the cipher option. * Updated the cipher option online XML documentation to specify what is expected for the value. * Updated pjsip.conf.sample to not indicate that ALL is acceptable since ALL does not imply a preference order for the ciphers and PJSIP does not simply pass the string to OpenSSL for interpretation. ASTERISK-24199 #close Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/4018/ ........ Merged revisions 424393 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424394 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	9ff743e995	Alembic: Add enumerator value to sippeers -> directmedia - 'outgoing' The 'outgoing' value was left off of the enumerator when first creating the column. This patch adds it, and should gracefully upgrade keeping the existing data in tact. ASTERISK-23781 #close Reported by: Stephen More Review: https://reviewboard.asterisk.org/r/4013/ ........ Merged revisions 424372 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	1e620fe59e	res_pjsip: document use of rewrite_contact in sample conf Without setting rewrite_contact, an invite to an endpoint behind NAT will not reach it - unless the endpoint itself uses STUN or TURN to discover it's public URI. Thus, the use of this should be in the sample documentation. Review: https://reviewboard.asterisk.org/r/4036/ ........ Merged revisions 424337 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	2dfc3b65f8	chan_pjsip: Fix an assertion for channels that lack formats on creation ASTERISK-24222 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4017/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	a2c47caa09	res_hep: Release allocation reference to configuration. ASTERISK-24362 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4026/ ........ Merged revisions 424312 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	a1763a89a3	res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash. During the latest update to DTLS-SRTP support the ability to configure the hash used for fingerprints was added. This gave us two supported ones: SHA-1 and SHA-256. The default was accordingly updated to SHA-256. Unfortunately this configuration ability was not exposed within res_pjsip. This change adds a dtls_fingerprint option that controls it. #SIPit31 ........ Merged revisions 424290 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	0de2d080c2	res_pjsip_sdp_rtp: Accept DTLS attributes in top level, not just media session. #SIPit31 ........ Merged revisions 424287 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	e3da76a352	PJSIP: Handle defaults properly This updates the code behind PJSIP configuration options with custom handlers to deal with the assigned default values properly where it makes sense and adjusting the default value where it doesn't. Before applying this patch, there were several cases where the default value for an option would prevent that config section from loading properly. Reported by: Thomas Thompson Review: https://reviewboard.asterisk.org/r/4019/ ........ Merged revisions 424263 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	ac095304e6	PJSIP: Force transport on contact rewrite If contact rewriting is enabled but the contact differs in transport from what is actually being used, messages after the initial INVITE transaction can be sent to an incorrect transport/port combination. In the case where this bug occurred the remote party never received a BYE since it was sent to the remote party's TCP port over UDP. Review: https://reviewboard.asterisk.org/r/4032/ ........ Merged revisions 424244 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	303547231e	chan_sip: Simplify some unref code by removing unlink_peer_from_tables. ASTERISK-22945 #related Reported by: ibercom Patches: asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License #6599) ........ Merged revisions 424181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424182 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424183 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	45e32e4b8c	chan_sip: Remove excess ref of realtime peer before sip_poke_peer. The peer is referenced at the end of sip_poke_peer, it should not get an extra ref before the call to sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close Reported by: ibercom Tested by: Yuriy Gorlichenko Patches: asterisk11.patch uploaded by ibercom (License #6599) Review: https://reviewboard.asterisk.org/r/4031/ ........ Merged revisions 424176 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424177 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424178 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d9b15388b2	res_pjsip_sdp_rtp: Don't place an extra whitespace before 'rport' and don't put IPv6 addresses in brackets. #SIPit31 ........ Merged revisions 424155 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	b1bb6b97df	res_rtp_asterisk: Ensure that the base and mapped address for candidates is present in SDP. This change fixes an issue where ICE candidates put into the SDP did not contain the 'raddr' and 'rport' information for server reflexive and relay candidates. #SIPit31 ........ Merged revisions 424151 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424152 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	faae530006	pjsip_cli: Suppress header print on error or no objects If there's an error on the pjsip command line or there are no objects, don't print the column headers. ASTERISK-24350 #close Reported-by: Brad Latus Tested-by: George Joseph Tested-by: Brad Latus Review: https://reviewboard.asterisk.org/r/4025/ ........ Merged revisions 424128 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	8d55892df7	autosupport: Fix bashism. '==' is bashism (bashspecific, fails when dash is /bin/sh). Anyway, a 'case' works better there. Originally committed in r375059 and r375060 on 2012-10-16 21:13:08. ASTERISK-20567 #close Reported by: Tzafrir Cohen ........ Merged revisions 424117 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424125 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	2a7c5208ee	Simplify UUID generation in several places. Replace code using ast_uuid_generate() with simpler and faster code using ast_uuid_generate_str(). The new code avoids a malloc(), free(), and copy. ........ Merged revisions 424103 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	00207158e1	threadpool.c: Minor cleanup fixes. * Fix threadpool_alloc() prototype. * Add missing off-nominal NULL check of pool in threadpool_alloc(). * searializer_create() does not need to create the object with a lock as the lock is not used. ........ Merged revisions 424096 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	19ffbb1e64	res_pjsip_session: Add additional checks for delaying session refreshes. There are certain situations which no checks existed for which need to prevent session refreshes. This includes sending a session refresh with SDP before SDP negotiation has completed and sending a session refresh before the dialog itself has been established. Checks for these have been added. Additionally COLP related UPDATEs were including SDP when it is not needed. Review: https://reviewboard.asterisk.org/r/4008/ ........ Merged revisions 424056 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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